CN102714830B - Method and device for handing over voice over internet protocol (VoIP) call from packet switch domain to circuit switch domain - Google Patents

Abstract

A method for handing over a voice over internet protocol (VoIP) call from packet switch (PS) domain to circuit switch (CS) domain in a communication system is provided. The communication system includes a VoIP session anchor device, a CS control device of the CS domain and a PS control device of the PS domain. When the CS control device receives a handover request from the PS control device, the method includes the following steps: the first time period is computed, wherein the first time period is the required time for the CS control device sending a message to the remote terminal of the VoIP call; the first time period and the preset second time period are compared, wherein the second time period is the required time for the CS control device sending a message to the local terminal of the VoIP call; according to the comparison result, sending a session handover request to the VoIP session anchor device and sending a cell handover request to the PS control device are controlled, enabling the session handover of IP multimedia subsystem (IMS) layer and the cell handover of layer 2 to occur simultaneously or nearly simultaneously. The present invention also relates to a device for handing over a VoIP call from the PS domain to the CS domain.

Description

Method and apparatus for switching a VoIP call from a packet-switched domain to a circuit-switched domain

technical field

The invention relates to a communication network, in particular to a method and a device for switching a VoIP call from a packet switching domain to a circuit switching domain.

Background technique

IMS is a global, access-independent and standard-based IP channel and service control system, which makes it possible for terminal users based on common Internet protocols to use different types of multimedia services. The IMS system not only provides multiple access methods, but also provides interoperability with the circuit switching domain (CS domain). For a multi-mode mobile terminal in the packet switched domain (PS domain), when it moves to the edge or outside of the network covered by the PS domain, or the network covered by the current PS domain becomes unavailable and other mobile terminals covered by the CS domain When in the network, the call can be switched from the PS domain to the CS domain to ensure the call quality.

In the prior art, if the user terminal can simultaneously receive and send information via the PS domain and the CS domain, the handover is usually completed before disconnecting the current connection. That is to say, the user terminal creates a call or session in the CS domain while maintaining the existing voice call in the PS domain. After the call or session in the telephone switching domain is established, the voice media is switched to the call or session in the CS domain. , and then release the network resources in the PS domain before the handover. This approach usually has minimal disruption. However, the above method cannot be used for a terminal that cannot simultaneously receive and transmit information through two domains.

Therefore, 3GPP TS 23.216 stipulates that calls anchored in the IMS system should be connected between EPS (Evolved Packet System) PS access and UTRAN/GERAN (Universal Terrestrial Radio Access Network/GSM EDGE Radio Access Network) CS access. The SRVCC (Single Radio Frequency Voice Call Continuity) solution between the two is used to maintain the continuity of the voice call. In this case, the UE (User Equipment) can only pass through the aforementioned two interfaces at a given A type of access to a network to send or receive data.

However, the SRVCC scheme stipulated in 3GPP TS 23.216 also has some defects, such as undeterminable voice interruption time, complicated signaling process, etc. The reason why there is an uncertain voice interruption time will be analyzed in detail below.

Figure 1 exemplarily shows the network architecture of the SRVCC where the VoIP call specified in 3GPP TS 23.216 is handed over from E-UTRAN (Evolved Universal Terrestrial Radio Access Network) to UTRAN/GERAN.

As shown in Figure 1, UE accesses IMS through E-UTRAN, S-GW/PDN GW. E-UTRAN, also known as LTE (Long Term Evolution Technology), includes several E-Node Bs, which are responsible for the radio access network. EPS integrates the functions of NodeB, RNC and CN of the existing WCDMA and TD-SCDMA systems, and simplifies them into two network units, eNodeB and EPC. EPC includes: MME (Mobility Management Entity), acting as a control node, responsible for signaling processing of the core network; S-GW (Serving Gateway)/PDN-GW (Packet Data Network Gateway), responsible for data processing of the core network. Among them, the non-3GPP radio access network can access the EPC through the PDN-GW, and the 3GPP radio access network can access the EPC through the S-GW.

In the environment shown in Figure 1, when the UE is at the coverage edge or outside the coverage area of E-UTRAN, it may decide to switch to the CS domain provided by UTRAN/GERAN. In UTRAN/GERAN, UE accesses the IMS network via the base station and MSC (Mobile Switching Center) server.

UTRAN has become a more important access mode of UMTS at present; GERAN is a key part of GSM formulated and maintained by 3GPP, and is also included in UMTS/GSM network. . Typically, a mobile operator's network consists of multiple GERANs, and in UMTS/GSM networks it is combined with UTRAN.

Please refer to TS 23.216 for details about other network elements in Figure 1 and communication methods between network elements.

Figure 2 shows the relevant call flow diagram of SRVCC handover from E-UTRAN to UTRAN/GERAN under the conditions stipulated in 3GPP TS 23.216. In order to complete the handover of the voice session, the voice call needs to be anchored in the IMS, such as the SCC AS (Service Centralization and Continuity Application Server) in advance. Figure 2 corresponds to Figure 6.2.2.1-1 of 3GPP TS 23.216.

As shown in Figure 2, when the source E-UTRAN decides to switch the ongoing VoIP call of the local UE from the PS domain to the CS domain according to the measurement report received from the local/source UE, it sends a handover request to the local MME, namely Step 203. Then the source MME divides the bearer in step 204 (for the subsequent transfer of the voice service), and in step 205 sends the corresponding handover from the PS domain to the CS domain to the MSC server/media gateway currently able to cover the local UE ask. After the corresponding MSC/media gateway prepares for the handover and establishes the circuit, it initiates the session transfer. Here, it should be noted that if the target MSC to which the local UE is to be handed over is the same as the MSC that received the handover request from the MME, the steps 206, 208, and 209 in the dotted line can be omitted (steps 220, 221 are also in this way).

Next, SRVCC includes a session handover process at the IMS layer and a cell handover process at layer 2 to the target cell. That is, it includes two user-level switches:

1) In steps 210 to 212, the SCC AS in the IMS executes the session handover process, updates the remote UE (that is, the party that establishes a VoIP session with the local UE) with the SDP of the target CS access leg, and releases the source EPC PS Access the legs. The above steps will result in the translation of the voice part of the ongoing session from the EPC to the MGW at the user level. In Fig. 2, the INVITE message in step 210 is a session handover request, because it contains the SDP information of the MGW, and the remote UE starts the session handover process after receiving the INVITE message via the SCC AS.

2) Steps 215-221, handover from E-UTRAN to UTRAN/GERAN performed at the local UE and the access network, which is a handover between RATs (radio access types) performed at the local UE and the access network , will cause the local UE to switch from the current E-UTRAN cell to the target UTRAN/GERAN cell.

During the above two switching processes, the VoIP call will be interrupted, that is, the voice stream will be interrupted. Although the steps are numbered with consecutive numbers in FIG. 2 , this does not indicate the temporal relationship between steps 210 to 212 and steps 215 to 221 . In fact, on the contrary, it is entirely possible that steps 210 to 212 are executed after step 215, or may be executed at the same time. That is to say, there is no synchronization mechanism between the two switching processes, which makes the time of voice interruption uncontrollable and uncertain. In the worst case, the voice interruption time will be very long, making the user experience worse.

Fig. 3 exemplarily shows the duration of the interruption of the VoIP call in the above SRVCC solution. Wherein, T1 is the interruption time of the layer 2 cell handover process, and also shows the start time and end time of the interruption. T2 is the interruption time of the session handover process at the IMS layer, and also shows the start time and end time of the interruption.

Case 1: The interruption caused by the session handover process at the IMS layer occurs earlier than the interruption caused by the cell handover process. In this case, the interrupt time generated by SRVCC will be greater than the maximum of T1 and T2, and in the worst case, equal to T1+T2.

Case 2: The interruption caused by the session switching process of the IMS layer and the interruption caused by the cell switching process occur at the same time. In this case, the interrupt time generated by SRVCC is equal to the maximum value of T1 and T2.

Case 3: The interruption caused by the cell switching process occurs earlier than the interruption caused by the session switching process at the IMS layer. In this case, similar to case 1, the interrupt time generated by SRVCC will be greater than the maximum value of T1 and T2, and in the worst case, equal to T1+T2.

Based on the above analysis, it can be seen that the voice interruption in the existing SRVCC scheme is mainly caused by the session handover process at the IMS layer and the cell handover process to the target cell at layer 2 (handover performed between the local UE and the access network). ) are generated in two parts. However, due to the arbitrariness of the execution of these two processes, the time of speech interruption becomes uncontrollable and uncertain. Prolonged voice interruption will undoubtedly deteriorate the user experience.

Contents of the invention

In order to solve the defects in the prior art, the present invention proposes to introduce a synchronization mechanism between the session handover process of the IMS layer and the cell handover process of layer 2, so that the two handover processes start at the same time or almost at the same time, thus the entire SRVCC Interruption times are reduced to a minimal constant value, improving user experience.

It should be noted that the control device MSC server in the CS domain may support the service interface to the IMS, or may not support the IMS interface but only support the interface to the ISUP/BICC. In this application, the latter case is mainly considered, that is, the MSC server only supports the interface to ISUP/BICC. It can be briefly reviewed: ISUP (ISDN User Part) is a main protocol of the SS7/C7 signaling system, which defines protocols and procedures for establishing, managing and releasing trunk circuits. Voice and data calls over the telephone network (PSTN). The BICC protocol (Bearer-Independent Call Control Protocol) is a protocol formulated by the ITU-TSG11 group. It evolved from the ISUP protocol and is a control protocol for implementing calls that are not related to service bearers in the backbone network.

In order to achieve the above object, according to an embodiment of the present invention, a method for switching a VoIP call from a packet switched PS domain to a circuit switched CS domain in a communication system is provided, the communication system includes a VoIP session anchor device, a CS domain The CS control device of the PS domain and the PS control device of the PS domain, when the CS control device receives a handover request from the PS control device, the method includes:

calculating a first time period, the first time period being the time required for sending a message from the CS control device to the remote terminal of the VoIP call;

comparing the first time period with a preset second time period, the second time period being the time required for the CS control device to send a message to reach the local terminal of the VoIP call; and

According to the comparison result, the sending of the session handover request to the VoIP session anchoring device and the sending of the cell handover request to the PS control device are controlled.

In a preferred embodiment of the present invention, the communication system further includes an intermediate control device, wherein calculating the first period of time further includes:

The CS control device triggers an intermediate control device of the communication system to send a non-session handover request to the VoIP session anchor device; and

The CS control device receives a response message from the intermediate control device, the response message being when the VoIP session anchor device interacts with the remote terminal and the intermediate control device receives a reply from the VoIP session anchor device Generated after the message.

According to an embodiment of the present invention, calculating the first time period includes storing the time when the CS control device triggers the intermediate control device, storing the time when the CS control device receives a reply message from the intermediate control device, and storing the second A period of time is 0.4 to 0.6 times, preferably 0.5 times, the time length between these two moments.

According to an embodiment of the present invention, when the first time period is greater than the second time period, the CS control device sets a timeout time equal to the a timer for the difference between the first period and the second period; and after the timer expires, the CS control device sends a cell switching request to the PS control device.

According to an embodiment of the present invention, when the first time period is equal to the second time period, the CS control device starts sending a session switch request to the VoIP session anchor device, and at the same time sends a session switching request to the PS control device Send a cell handover request.

According to an embodiment of the present invention, when the first time period is less than the second time period, the CS control device sets a timeout time equal to the second time period while sending a cell switching request to the PS control device. A timer for the difference between the time period and the first time period; and after the timer expires, start sending a session switching request to the VoIP session anchoring device.

On the other hand, an embodiment of the present invention also provides a CS control device for switching a VoIP call from a packet-switched PS domain to a circuit-switched CS domain in a communication system, and the communication system further includes a VoIP session anchor device and a PS Domain PS control equipment, the CS control equipment includes:

Calculation means for calculating a first period of time when receiving a switching request from the PS control device, the first period being the time required for a message sent from the CS control device to reach the remote terminal of the VoIP call;

Comparing means for comparing the first time period with a preset second time period, the second time period being the time required for the CS control device to send a message to reach the local terminal of the VoIP call; and

A control device, configured to control the sending of the session handover request to the VoIP session anchoring device and the sending of the cell handover request to the PS control device according to the comparison result.

According to an embodiment of the present invention, the control device is configured to: when the first time period is greater than the second time period, when starting to send a session switching request to the VoIP session anchoring device, set a timeout time equal to A timer for the difference between the first period and the second period; and sending a cell switching request to the PS control device after the timer expires.

According to an embodiment of the present invention, the control device is configured to: when the first time period and the second time period are equal, start sending a session switching request to the VoIP session anchoring device, and at the same time send a session switching request to the PS control device Send a cell handover request.

According to an embodiment of the present invention, the control device is configured to: when the first time period is less than the second time period, when sending a cell switching request to the PS control device, set a timeout time equal to the second time period A timer for the difference between the second time period and the first time period; and after the timer expires, start sending a session switching request to the VoIP session anchoring device.

On the other hand, the present invention also provides an intermediate control device in a communication system, and the communication system further includes a VoIP session anchor device, a PS control device in a PS domain, and a method for switching a VoIP call from a packet-switched PS domain to a circuit-switched The CS control device of the CS domain,

The intermediate control device includes an auxiliary computing device configured to: trigger by the CS control device, send a non-session switching request to the VoIP session anchor device; After interacting with the remote terminal of the VoIP session anchoring device, a reply message is received from the VoIP session anchor device; and, a reply message is sent to the CS control device, so that the CS control device calculates a first period, the first period being the CS Controls the time it takes for a device to send a message to the remote end of a VoIP call,

The control device also includes auxiliary control means configured to: at the CS control device compare the first time period with a second time period representing the time required for the CS control device to send a message to the local terminal of the VoIP call After a time period, the CS control device initiates sending a session switch request to the VoIP session anchor device.

By using the method and equipment of the present invention, a synchronization mechanism is introduced between the two main handover processes that cause voice interruption in the SRVCC scheme, that is, the session handover process at the IMS layer and the cell handover process at layer 2, so that the two handover processes Interruption occurs simultaneously or almost simultaneously, so that the interruption of the entire SRVCC process becomes controllable and constant, and is reduced to a minimum value, which improves system performance and obviously improves user service quality. Moreover, the solution of the present invention basically has no impact on the existing network environment, thus having high compatibility.

Description of drawings

Other characteristics, objects and advantages of the present invention will become more apparent by reading the following detailed description of non-limiting embodiments with reference to the accompanying drawings.

Figure 1 shows the network architecture of SRVCC specified in 3GPP TS 23.216;

FIG. 2 is a schematic diagram of a related SRVCC process for switching from E-UTRAN to target GERAN without DTM/PSHO support in the prior art;

FIG. 3 is a schematic diagram of SRVCC interruption time length in the prior art;

Figure 4 is a message sequence diagram according to a particular embodiment of the present invention;

Figure 5 is a flowchart according to an embodiment of the present invention; and

FIG. 6 is a functional block diagram of a CS control device and an intermediate control device according to an embodiment of the present invention.

Detailed ways

In Fig. 4, the local UE is about to switch from the PS domain to the CS domain. In the figure, the source MME is the control device responsible for processing signaling in the PS domain. The MSC server is a control device in the CS domain (including WCDMA, GSM, TD-SCDMA and other CS domains) responsible for processing signaling. Of course, the basic idea of the present invention can also be applied to the MSC server in the CDMA network. The SCC AS can be used as the VoIP session anchor device in the IMS. Of course, the VoIP session anchor device can also be an IBCF (Interconnect Border Control Function) or other logical entities. In one embodiment of the present invention, the MSC server only supports the interface to ISUP/BICC, and an intermediate control device is needed to complete the possible signaling processing tasks. The MGCF acts as an intermediate control device handling signaling in this embodiment. Correspondingly, IM-MGW (IP Multimedia Gateway) is used to handle the media codec conversion of the user plane.

In order to complete the VoIP handover mechanism proposed by the present invention, a value P2 needs to be preset in the MSC server, and this value indicates the time required for the MSC server to send a message of switching with CS to the local UE. Specifically, it represents the average time required for the MSC server to send the PS to CS Response message to the MME until the local UE finally receives the HO from EUTRAN Command. This average time is mainly composed of two parts: the sum of the processing time of the communication nodes through which the message passes and the time required for the message to be transmitted in the network. Because the number of nodes passed by the message to reach the UE is fixed, the time required for these nodes to process the message can be calculated. In addition, since the MSC server and the local UE are in the same local network, the distance from the MSC server to the UE in the local network does not change much, so the time required for the message to transmit in the network does not change much. Estimated by experience. It can be seen that the time required for the message to be sent from the MSC server to the final arrival at the local UE can be calculated and estimated based on experience, therefore, P2 can be preset.

Fig. 4 shows a sequence diagram of signaling messages according to the present invention. It should be pointed out that although the various steps in the figure are numbered with consecutive numbers, these do not indicate the order in which the various steps are executed.

Steps 401 to 409 are similar to steps 201 to 209 in FIG. 2 , and will not be described here.

In step 409a, when the MSC server detects that SRVCC handover is to be performed, it sends a BICC IAM message with STN-SR (session transfer number-single radio frequency) to the MGCF. In this embodiment, the IAM message is used to trigger the MGCF to perform subsequent operations. In addition, the MSC server will store the time when the IAM message is sent, for example mark this time as T4.

In step 410, upon receiving the IAM message including STN-SR, the MGCF detects that this is an SRVCC procedure. Then the MGCF sends an empty INVITE request not including the SDP information of the IM-MGW to the S-CSCF in the IMS, and the S-CSCF forwards the empty INVITE request to the VoIP session anchor device according to standard procedures, that is, the SDP in this embodiment SCC AS. It should be noted that the empty INVITE request does not include the SDP information of the IM-MGW. In other words, the message is not a session switch request in the traditional sense, and we may call it a "non-session switch request" to distinguish it from a session switch request in the general sense. Of course, other messages used to measure the time required for the MSC to send a message to the remote UE are also possible.

In step 410a, after receiving the above INVITE message, the SCC AS interacts with the remote UE. The interactive process may include: based on the INVITE request in step 410, the SCC AS forwards the re-INVITE request without SDP information to the remote UE. After the remote UE processes the request, it responds to the SCC AS with a 200 OK message with its own SDP information.

In step 410b, the MGCF sends an ACM message to the MSC server before the response message ANM is sent to the MSC server. Here we need to briefly recall that in the CS domain, the ACM message refers to address completion information, indicating that the remote terminal of the trunk circuit has been reserved. ANM is an answer message, indicating that the called party has picked up the phone to answer.

There are several ways of determining how to send an ACM message. In this embodiment, the following two methods can be used: MGCF sends BICC ACM message to MSC server after the timer defined in Section 7.2.3.2.4 of TS 29.163 expires. Another option is that when receiving the final successful response (for example, 200OK response), the MGCF sends an ACM message and a BICC ANM message to the MSC server. Of course, other methods can also be used.

In step 411, after communicating with the remote UE, the SCC AS responds to the MGCF with a 200 OK message with the remote UE's SDP information. Since the remote UE has not obtained the SDP information of the IM-MGW, the handover process of the remote UE cannot be started, and so far, the media stream of the current session remains connected.

In step 411a, the MGCF sends an ANM message to the MSC server as defined in TS 29.163. When the MSC receives the ANM message, it stores this time and may mark this time as T5. At that moment, the media stream of the ongoing session is still connected. Since in step 410, the MGCF has detected that this is an SRVCC process, it does not send an ACK message to the SCC AS at this time.

After step 411a, once the MSC server stores and marks T5, it will calculate the duration taken from the MSC server to the remote UE from T4 and T5 (this duration can be denoted as P1 for example). In this embodiment, the BICC message and the SIP message are sent and received during the period from T4 to T5 (step 409a to step 411a), and the MSC server can calculate the time required for sending the message to the remote UE according to the round trip of the above message. Time P1, for example, P1 is calculated as a certain ratio of the time length between time T4 and time T5, and this ratio can be set according to experience and network conditions. In the case of symmetrical round trips, this ratio can be 0.5. A ratio of 0.55. or 0.45 is also possible to account for possible inconsistencies in the round trip. Generally, the ratio can be set between 0.4 and 0.6 and adjusted according to the actual situation. As mentioned above, the value of the duration P2 during which the MSC server sends the CS handover-related message to the local UE can be preset by the operator. After calculating P1, the MSC server will compare P1 with the preset size of P2. According to different comparison results, the MSC server will perform different actions.

The steps of calculating P1 by the MSC server and comparing the sizes of P1 and P2 are not clearly marked in FIG. 4 , but it should be understood that the steps of calculating and comparing are performed immediately after step 411a. This is an extremely simple algorithm that hardly takes up additional resources of MSC.

Steps 411b and 413 will be discussed in detail below: when the MSC server receives the ANM message from the MGCF, after multiple calculations and comparisons, the MSC will decide whether to set a timer according to the comparison result. For example, when the sizes of P1 and P2 are different, the MSC server will set a timer, and the timeout value of the timer can be set as the difference between P1 and P2.

Example 1: P1 is greater than P2, indicating that the time required by the MSC server to send a message to a remote UE is greater than the time it takes to send a message to a local UE. In this case, for the MSC server, the session switching at the IMS layer should be started first, for example, it can start the MGCF to send a session switching request to the SCC AS.

In the case of Example 1, the MSC server will execute step 411b and step 413 at the same time. On the one hand, the MSC server resends the same initial IAM message as the IAM message in step 409a to the MGCF, and we can mark this IAM message as re-IAM to distinguish it from the message in step 409a. On the other hand, the MSC server will start a timer at the same time, and the timeout period of the timer is P1-P2.

Initiated or driven by the re-IAM message sent by the MSC server, as shown in step 412, the MGCF sends an ACK message including the IM-MGW to the SCC AS of the IMS, which is an actual session handover request message. In steps 417 and 418, after the SCC AS receives the ACK message of the SDP with the MGW, it will immediately send an ACK message to the remote UE based on the ACK message to start the session handover process and update the remote UE. The subsequent process and the existing Similar to that defined in the 3GPPTS 23.216 specification.

Note that after the timer set in step 413 expires, the MSC server sends a cell handover request to the source MME, such as the PS to CS Response message shown in step 14 in Figure 4, and then executes steps 15 and 16 to start the local The cell handover process of the UE side handover to the target cell, the subsequent process is similar to that defined in the 3GPP TS 23.216 specification, and will not be described in detail here, so as not to obscure the present invention unnecessarily.

Example 2: If P1 is equal to P2, it means that the time required by the MSC server to send a message to a remote UE is the same as the time required to send a message to a local UE. In this case, the MSC server does not need to set a timer, and performs step 411b and step 414 at the same time. On the one hand, the MSC server drives or starts the MGCF to send a real session switching request to the SCC AS by sending the same initial IAM message as the IAM message in step 409a again, that is, the re-IAM message; on the other hand, as shown in step 414 As indicated, the MSC server sends a PS to CS Response message to the source MME. In other words, in Figure 4, the re-IAM message and the PS to CSResponse message are sent at the same time.

And in the following step 412, when the re-IAM message is received, the MGCF finds that the call is still going on and related to the SRVCC process. Therefore, the MGCF discards the re-IAM message and sends an ACK message with the SDP information of the IM-MGW to the SCC AS. Certainly, the SDP information should be consistent with the SDP information in the 200OK received from the SCC AS/remote UE in step 411. After the SCC AS receives the above ACK message, it will immediately send an ACK message to the remote UE based on the ACK message to start the session handover process and update the remote UE, and the source MME will start the local UE side after receiving the PS to CS Response message. Cell handover process.

Example 3: P2 is greater than P1, that is, the time required for the MSC server to send a message to the local UE is longer than the time required for it to send a message to the remote UE, the MSC server should first send a cell handover request to the source MME, and then start the session handover process. Therefore, the timeout period of the timer will be set as P2-P1.

In step 414 in FIG. 4, the MSC server sends a PS to CSResponse message to the source MME to start the cell handover process on the local UE side. While step 414 is being executed, the MSC server starts a timer (that is, step 413), and the timeout period of the timer is P2-P1. Only after the timer expires, the MSC server starts the session switching of the IMS layer, for example, sends a re-IAM message to the MGCF to start the session switching, ie step 411b. In other words, if P2 is greater than P1, step 411b will be performed after step 414, the timer is started at step 414, and step 411b will be performed only after the timer expires.

For ease of understanding, FIG. 5 shows a flow chart of the method corresponding to the signaling message sequence diagram in FIG. 4 .

As shown in the figure, after receiving the PS to CS req message sent by the source MME in step 405, the MSC server calculates P1 through steps 409a to 411a in Figure 4, that is, it takes the MSC server to send a signaling message to the remote UE duration, and after step 411a and before other possible steps (steps 411b, 413 or 414), the process of comparing P1 with the preset value P2 is started. The comparison produces three possible results. When P1>P2, steps 411b and 413 will be executed at the same time, that is, a timer with a timeout period of P1-P2 will be set while session switching is started. After the timer expires, step 414 is executed, that is, the cell handover on the local UE side is started by sending PS to CS Req to the source MME. When P1=P2, no timer needs to be set, and steps 411b and 414 are executed simultaneously. When P1<P2, steps 414 and 413 are performed at the same time, that is, a timer with a timeout period of P2-P1 is set while starting the cell handover on the local UE side. When the timer expires, step 411b is executed, for example, by sending a re-IAM message to the MGCF to initiate session switching.

It should be noted that the purpose of step 409a to step 411a is to calculate P1 instead of actually starting the session handover at the IMS layer. This is fundamentally different from step 210 in Figure 2 (actually starting the session handover at the IMS layer). In addition, the method for calculating P1 is not limited to the above INVITE-200OK message pair, and other messages, such as INVITE-183 provisional response message, may also be used.

It is worth noting that in the embodiment described with reference to FIG. 4, the MSC server does not support the SIP interface. In other embodiments of the present invention, when the MSC server supports the SIP interface, the flow shown in FIG. 5 can still be used to reduce the interruption time of the entire SRVCC process. For example, the value of P1 can be calculated by the MSC server sending an INVITE message not including the IM-MGW to the SCC AS, receiving a 200OK message with the SDP of the remote UE from the SCC AS, and sending a message to start the session handover of the IMS layer by the MSC server ask. In such cases, the MGCF entity is not required.

Fig. 6 shows a functional block diagram of a CS control device 601 according to an embodiment of the present invention and an intermediate control device 602 according to an embodiment of the present invention, taking an MSC server and an MGCF device as examples.

In FIG. 6 , the MSC server includes: computing means 611 , comparing means 612 and controlling means 613 . The MGCF device 602 includes an auxiliary computing device 621 and an auxiliary control device 622 . It is worth noting that the presence of the MGCF equipment is not essential, so the interaction between the computing means 611 and the auxiliary computing means 621 and the interaction between the control means 613 and the auxiliary control means 622 are shown with dashed lines. It is worth noting that the above equipment includes more than the functional modules shown in Figure 6, for example, it also includes various message receiving and sending units, and only the parts related to the embodiment of the present invention are shown in the figure to avoid unnecessary Obscure the invention. Various devices in the figure can be realized by computer program codes on the basis of existing functional entities.

When the calculation means 611 receives the handover request (PS to CS Req) from the PS control equipment (such as the source MME), it calculates the time P1 required for sending a message from the CS control equipment (ie, the MSC server) to the remote terminal of the VoIP call.

In the case that the MSC server supports the SIP interface, in order to "measure" the elapsed time P1 from the MSC server sending a message to the remote terminal, the computing device 611 can be configured to: send an empty INVITE message to the VoIP session anchoring device SCC AS, that is, not For session switching requests and for receiving response messages from the SCC AS after the SCC AS has interacted with the remote terminal, half or approximately half of the time period elapsed between sending and receiving the above messages is taken as the value of P1.

In the case that the MSC server does not support the SIP interface, in order to "measure" the elapsed time P1 from the MSC server sending a message to the remote terminal, the computing device 611, for example, triggers the auxiliary computing device 621 to send a non-session switching request to the SCC AS through an IAM message, The auxiliary computing device 621 receives a response message such as 200 OK from the SCC AS after the interaction between the SCC AS and the remote terminal, and returns a response message such as ANM to the MSC 601. The MSC calculates P1 according to the time of "trigger" and the time of receiving the reply message.

The comparing means 612 compares P1 with the preset P2. As mentioned above, P2 is the time required for the message sent by the MSC server to reach the local terminal of the VoIP call.

The control means 613 will control the start time of session handover and cell handover according to the comparison result of P1 and P2.

A timer 614 is also shown in block 601 in FIG. 6 , and it should be noted that the timer 614 is also not necessary. When P1 is the same as P2, the MSC server 601 does not need to set a timer, and starts session handover and cell handover of the local UE at the same time. Only when P1 and P2 are different, set a timer whose timeout time is equal to |P1-P2|. When P1>P2, the control device 613 sets a timer while starting the session switching, and starts the cell switching only after the timer expires. When P2>P1, the control device 613 starts the cell handover and sets a timer at the same time, and starts the session handover only after the timer expires.

In case the MSC server supports the SIP interface, the MSC server 601 initiates session handover at the IMS layer, for example by sending a session handover request to the SCC AS, for example, by sending a cell handover request to the PS control device (e.g., source MME) .

In the case that the MSC server does not support the SIP interface, the control means 613 initiates session switching by sending a re-IAM message to the auxiliary control means 622, for example.

The specific embodiments of the present invention have been described above with reference to the accompanying drawings. It should be understood that the present invention is not limited to the specific embodiments described above, and those skilled in the art may make various variations or modifications within the scope of the appended claims.

Claims (11)

1. one kind is switched to voip call the method in circuit switching (CS) territory in communication system from packet switching PS territory, described communication system comprises the CS control appliance in voip conversation anchoring arrangement, CS territory and the PS control appliance in PS territory, in the time that described CS control appliance receives the handover request from described PS control appliance, the method comprises:

Calculated for the first period, described the first period is to send a message to from described CS control appliance the required time of remote terminal that reaches voip call;

More described the first period and predefined the second period, described the second period is that described CS control appliance sends a message to the required time of local terminal that reaches voip call; And

According to comparative result, control to described voip conversation anchoring arrangement session handover request transmission and to the transmission of the community handover request of described PS control appliance.

2. the method for claim 1, also comprises neutral-controlled plant, wherein calculates for the first period further to comprise:

The neutral-controlled plant that described CS control appliance triggers described communication system sends non-session handover request to described voip conversation anchoring arrangement; And

Described CS control appliance receives response message from described neutral-controlled plant, and described response message is at described voip conversation anchoring arrangement and described remote terminal is mutual and described neutral-controlled plant generates from message is replied in described voip conversation anchoring arrangement reception.

3. method as claimed in claim 2, wherein calculate the moment that the first period comprised that the described CS control appliance of storage triggered described neutral-controlled plant, store the moment of described CS control appliance from described neutral-controlled plant reception response message, and be 0.4 to 0.6 times of time span between these two moment by described the first period value, be preferably 0.5 times.

4. the method as described in claims 1 to 3 any one, further comprise: in the situation that described the first period is greater than described the second period, described CS control appliance, in starting and sending session handover request to described voip conversation anchoring arrangement, arranges time-out time and equals the timer of the difference of described the first period and described the second period; And after described timer expiry, described CS control appliance sends community handover request to described PS control appliance.

5. the method as described in claims 1 to 3 any one, further comprise: in the situation that described the first period equals described the second period, described CS control appliance starts to described voip conversation anchoring arrangement transmission session handover request, and sends community handover request to described PS control appliance simultaneously.

6. the method as described in claims 1 to 3 any one, further comprise: in the situation that described the first period is less than described the second period, described CS control appliance, in sending community handover request to described PS control appliance, arranges time-out time and equals the timer of the difference of described the second period and described the first period; And after described timer expiry, start to described voip conversation anchoring arrangement and send session handover request.

7. a CS control appliance that voip call is switched to circuit switching (CS) territory in communication system from packet switching PS territory, described communication system also comprises the PS control appliance in voip conversation anchoring arrangement and PS territory, described CS control appliance comprises:

Calculation element, for when the handover request receiving from described PS control appliance, calculated for the first period, and described the first period is to send a message to from described CS control appliance the required time of remote terminal that reaches voip call;

Comparison means, for more described the first period and predefined the second period, described the second period is that described CS control appliance sends a message to the required time of local terminal that reaches voip call; And

Control device, for control to according to comparative result described voip conversation anchoring arrangement session handover request transmission and to the transmission of the community handover request of described PS control appliance.

8. CS control appliance as claimed in claim 7, wherein said control device is configured to: in the time that described the first period is greater than described the second period, in starting and sending session handover request to described voip conversation anchoring arrangement, time-out time be set equal the timer of the difference of described the first period and described the second period; And after described timer expiry, send community handover request to described PS control appliance.

9. CS control appliance as claimed in claim 7, wherein said control device is configured to: in the time that described the first period is equal with the second period, start to described voip conversation anchoring arrangement and send session handover request, and send community handover request to described PS control appliance simultaneously.

10. CS control appliance as claimed in claim 7, wherein said control device is configured to: in the time that described the first period is less than described the second period, in sending community handover request to described PS control appliance, time-out time be set equal the timer of the difference of described the second period and described the first period; And after described timer expiry, start to described voip conversation anchoring arrangement and send session handover request.

Neutral-controlled plant in 11. 1 kinds of communication systems, described communication system also comprises the PS control appliance in voip conversation anchoring arrangement, PS territory and voip call is switched to the CS control appliance in circuit switching (CS) territory from packet switching PS territory,

Described neutral-controlled plant comprises auxiliary calculation element, and this device is configured to: by the triggering of described CS control appliance, send non-session handover request to described voip conversation anchoring arrangement; After the remote terminal of described voip conversation anchoring arrangement and voip call is mutual, receives and reply message from described voip conversation anchoring arrangement; And, send response message to described CS control appliance, so that described CS control appliance calculated for the first period, described the first period is that described CS control appliance sends a message to the required time of remote terminal that reaches voip call,

Described control appliance also comprises sub controlling unit, this device is configured to: more described the first period of described CS control appliance with represent that described CS control appliance sends a message to and reaches the local terminal of voip call the second period of required time after, start to described voip conversation anchoring arrangement and send session handover request through described CS control appliance.