Method and system for implementing multi-channel voice calls based on the same account

Abstract

The present application relates to a method and system for realizing multi-channel voice calls based on the same account, and relates to the field of communication voice technology, including establishing call connections between multiple PBX terminals and corresponding NGN terminals using the same VoIP account; allocating voice stream time slots for the connection between each PBX terminal and its corresponding NGN terminal based on the call connection information corresponding to each call connection; and performing voice stream conversion and communication between each PBX terminal and its corresponding NGN terminal based on the allocated voice stream time slots. The present application creates a voice call connection for each PBX user and its corresponding NGN terminal user sharing the same VoIP account, and allocates voice stream time slots for them, so that each PBX user and its corresponding NGN terminal user realize voice stream conversion and communication based on the voice stream time slot, thereby realizing successful access of multi-channel voices of the same account under the PBX.

Description

Technical Field

[0001] The present application relates to the field of communication voice technology, and in particular to a method and system for implementing multi-channel voice calls based on the same account. Background Art

[0002] With the popularization of broadband and the rapid development of VoIP (Voice over Internet Protocol, voice transmission based on IP) network technology, traditional PBX (Private Branch Exchange, user-level switch) services can no longer meet the needs of corporate customers, and more and more companies are using VoIP networks to reduce operating costs. Therefore, it has become a trend to connect VoIP networks to PBXs used for voice communication without changing the existing network architecture. Among them, enterprises make full use of existing network resources to deploy their own VoIP systems. Without changing the PBX service functions, they connect PBXs to VoIP networks, which not only saves users' long-distance call charges, but also expands voice communications with VoIP users, increasing the diversity of corporate voice services.

[0003] At present, ISDN PRI (Primary Rate Interface) users of enterprise PBXs, considering resource utilization and cost issues, hope to smoothly connect ISDN PRI users under PBX to NGN (Next Generation Network) network without changing the original ISDN (Integrated Services Digital Network) network, and without changing the existing PBX service functions. However, due to the particularity of PBX, that is, it requires that multiple internal extension users can use the same PBX external line account to talk to different external personnel at the same time; therefore, when PBX is integrated into NGN, there is a problem of accessing multiple voice channels under the same PBX external line account under PBX. Summary of the invention

[0004] The present application provides a method and system for implementing multi-channel voice calls based on the same account, so as to solve the problem in the related art that when the PBX is integrated into the NGN end, the access of multi-channel voices of the same account under the PBX cannot be implemented.

[0005] In a first aspect, a method for implementing multi-way voice calls based on the same account is provided, comprising the following steps:

[0006] Establish call connections between multiple PBX terminals and corresponding NGN terminals using the same VoIP account;

[0007] Allocate a voice stream time slot for the connection between each PBX end and its corresponding NGN end based on the call connection information corresponding to each call connection;

[0008] Based on the allocated voice stream time slot, voice stream conversion and communication are performed between each PBX end and its corresponding NGN end.

[0009] In some embodiments, when the PBX terminal is used as a calling terminal and the NGN terminal is used as a called terminal, the step of establishing a call connection between multiple PBX terminals using the same VoIP account and corresponding NGN terminals includes:

[0010] The ISDN control module receives and parses the ISDN SETUP signaling sent by the first PBX terminal, obtains the call information of the first ISDN voice, the call information includes the ISDN calling number, the VoIP called number and the ISDN time slot, and creates a first ISDN call unit for processing the first ISDN voice according to the ISDN calling number and the ISDN time slot;

[0011] The first ISDN calling unit converts the ISDN calling number into a VoIP calling number, and sends the VoIP calling number, the VoIP called number and the ISDN time slot to the VoIP protocol module;

[0012] The VoIP protocol module creates a first VoIP signaling unit for processing a first ISDN voice according to the VoIP calling number and the ISDN time slot;

[0013] The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP calling number, sends the VoIP calling number, the VoIP called number and the calling end SDP message to the NGN end corresponding to the first PBX end, and obtains the response signaling sent by the NGN end according to the VoIP calling number, the VoIP called number and the calling end SDP message, wherein the response signaling includes the called end SDP message;

[0014] The first VoIP signaling unit converts the response signaling into a call CONNECT message, and sends the call CONNECT message to the ISDN control module;

[0015] The first ISDN call unit in the ISDN control module constructs ISDN CONNECT signaling according to the call CONNECT message, and sends the ISDN CONNECT signaling to the first PBX end, so that the first PBX end establishes a call connection with its corresponding NGN end based on the ISDN CONNECT signaling.

[0016] In some embodiments, when the PBX terminal is used as the called terminal and the NGN terminal is used as the calling terminal, the step of establishing a call connection between multiple PBX terminals using the same VoIP account and corresponding NGN terminals includes:

[0017] The VoIP protocol module receives and parses the SIP INVITE signaling sent by the first NGN end, obtains the call information of the first ISDN voice, the call information includes the VoIP called number, the calling end SDP message and the signaling key value, and creates a first VoIP signaling unit for processing the first ISDN voice according to the VoIP called number and the signaling key value;

[0018] The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP called number, and sends the VoIP called number to the ISDN control module;

[0019] The ISDN control module converts the VoIP called number into an ISDN called number, allocates an ISDN time slot to the first ISDN voice according to the ISDN called number, and returns the ISDN time slot to the first VoIP signaling unit;

[0020] The first VoIP signaling unit converts the SIP INVITE signaling into a call SETUP message, and sends the call SETUP message to the ISDN control module;

[0021] The ISDN control module creates a first ISDN call unit for processing the first ISDN voice according to the ISDN called number and the ISDN time slot;

[0022] The first ISDN call unit constructs an ISDN SETUP signaling according to the call SETUP message, sends the ISDN SETUP signaling to the PBX end corresponding to the first NGN end, and obtains the ISDN CONNECT signaling sent by the PBX end according to the ISDN SETUP signaling;

[0023] The first ISDN call unit converts the ISDN CONNECT signaling into a call CONNECT message, and sends the call CONNECT message to the VoIP protocol module;

[0024] The first VoIP signaling unit in the VoIP protocol module constructs a response signaling according to the call CONNECT message and sends it to the first NGN end. The response signaling includes the called end SDP message, so that the first NGN end can establish a call connection with its corresponding PBX end based on the response signaling.

[0025] In some embodiments, allocating a voice stream time slot for a connection between each PBX end and its corresponding NGN end based on call connection information corresponding to each call connection includes:

[0026] The ISDN PRI access module receives the call connection information of the first ISDN voice sent by the first VoIP signaling unit, wherein the call connection information includes the PRI port, the ISDN time slot, the calling end SDP message and the called end SDP message;

[0027] The ISDN PRI access module creates a first voice stream unit for processing a first ISDN voice according to the PRI port and the ISDN time slot;

[0028] The first voice stream unit configures parameters of a certain idle DSP time slot according to the SDP message of the calling end and the SDP message of the called end to obtain the first idle DSP time slot, and connects the first idle DSP time slot with the ISDN time slot based on a preset time slot mapping relationship to form a voice stream time slot between the PBX end and its corresponding NGN end.

[0029] In some embodiments, the voice stream conversion and communication between the PBX calling terminal and its corresponding NGN called terminal based on the allocated voice stream time slot includes:

[0030] The first voice stream unit converts the PCM voice stream of the PBX end into a first VoIP voice stream based on the SDP message of the called end, and the first VoIP voice stream is transmitted to the NGN end through the voice stream time slot;

[0031] The first voice stream unit converts the VoIP voice stream of the NGN end into a first PCM voice stream based on the SDP message of the calling end, and the first PCM voice stream is transmitted to the PBX end through the voice stream time slot.

[0032] In a second aspect, a system for implementing multiple voice calls based on the same account is provided, including a VoIP gateway, wherein the VoIP gateway is used to establish call connections between multiple PBX terminals using the same VoIP account and corresponding NGN terminals; based on call connection information corresponding to each call connection, a voice stream time slot is allocated for the connection between each PBX terminal and its corresponding NGN terminal; based on the allocated voice stream time slot, voice stream conversion and communication are performed between each PBX terminal and its corresponding NGN terminal.

[0033] In some embodiments, when the PBX end serves as the calling end and the NGN end serves as the called end, the VoIP gateway includes an ISDN control module and a VoIP protocol module;

[0034] The ISDN control module receives and parses the ISDN SETUP signaling sent by the first PBX terminal, obtains the call information of the first ISDN voice, the call information includes the ISDN calling number, the VoIP called number and the ISDN time slot, and creates a first ISDN call unit for processing the first ISDN voice according to the ISDN calling number and the ISDN time slot;

[0035] The first ISDN calling unit converts the ISDN calling number into a VoIP calling number, and sends the VoIP calling number, the VoIP called number and the ISDN time slot to the VoIP protocol module;

[0036] The VoIP protocol module creates a first VoIP signaling unit for processing a first ISDN voice according to the VoIP calling number and the ISDN time slot;

[0037] The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP calling number, sends the VoIP calling number, the VoIP called number and the calling end SDP message to the NGN end corresponding to the first PBX end, and obtains the response signaling sent by the NGN end according to the VoIP calling number, the VoIP called number and the calling end SDP message, wherein the response signaling includes the called end SDP message;

[0038] The first VoIP signaling unit converts the response signaling into a call CONNECT message, and sends the call CONNECT message to the ISDN control module;

[0039] The first ISDN call unit in the ISDN control module constructs ISDN CONNECT signaling according to the call CONNECT message, and sends the ISDN CONNECT signaling to the first PBX end, so that the first PBX end establishes a call connection with its corresponding NGN end based on the ISDN CONNECT signaling.

[0040] In some embodiments, when the PBX end serves as the called end and the NGN end serves as the calling end, the VoIP gateway includes an ISDN control module and a VoIP protocol module;

[0041] The VoIP protocol module receives and parses the SIP INVITE signaling sent by the first NGN end, obtains the call information of the first ISDN voice, the call information includes the VoIP called number, the calling end SDP message and the signaling key value, and creates a first VoIP signaling unit for processing the first ISDN voice according to the VoIP called number and the signaling key value;

[0042] The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP called number, and sends the VoIP called number to the ISDN control module;

[0043] The ISDN control module converts the VoIP called number into an ISDN called number, allocates an ISDN time slot to the first ISDN voice according to the ISDN called number, and returns the ISDN time slot to the first VoIP signaling unit;

[0044] The first VoIP signaling unit converts the SIP INVITE signaling into a call SETUP message, and sends the call SETUP message to the ISDN control module;

[0045] The ISDN control module creates a first ISDN call unit for processing the first ISDN voice according to the ISDN called number and the ISDN time slot;

[0046] The first ISDN call unit constructs an ISDN SETUP signaling according to the call SETUP message, sends the ISDN SETUP signaling to the PBX end corresponding to the first NGN end, and obtains the ISDN CONNECT signaling sent by the PBX end according to the ISDN SETUP signaling;

[0047] The first ISDN call unit converts the ISDN CONNECT signaling into a call CONNECT message, and sends the call CONNECT message to the VoIP protocol module;

[0048] The first VoIP signaling unit in the VoIP protocol module constructs a response signaling according to the call CONNECT message and sends it to the first NGN end. The response signaling includes the called end SDP message, so that the first NGN end can establish a call connection with its corresponding PBX end based on the response signaling.

[0049] In some embodiments, the VoIP gateway further includes an ISDN PRI access module, and the ISDN PRI access module is specifically used to:

[0050] Receiving call connection information of a first ISDN voice sent by a first VoIP signaling unit, the call connection information including a PRI port, an ISDN time slot, a calling end SDP message, and a called end SDP message;

[0051] Creating a first voice stream unit for processing a first ISDN voice according to the PRI port and the ISDN time slot;

[0052] The first voice stream unit configures parameters of a certain idle DSP time slot according to the SDP message of the calling end and the SDP message of the called end to obtain the first idle DSP time slot, and connects the first idle DSP time slot with the ISDN time slot based on a preset time slot mapping relationship to form a voice stream time slot between the PBX end and its corresponding NGN end.

[0053] In some embodiments, the first voice stream unit is further used for:

[0054] Converting the PCM voice stream of the PBX end into a first VoIP voice stream based on the SDP message of the called end, wherein the first VoIP voice stream is transmitted to the NGN end through the voice stream time slot;

[0055] The VoIP voice stream of the NGN end is converted into a first PCM voice stream based on the SDP message of the calling end, and the first PCM voice stream is transmitted to the PBX end through the voice stream time slot.

[0056] The beneficial effects brought by the technical solution provided by this application include: successful access to multiple voice channels of the same account under a PBX can be achieved.

[0057] The present application provides a method and system for realizing multi-channel voice calls based on the same account, including establishing call connections between multiple PBX terminals using the same VoIP account and corresponding NGN terminals; allocating voice stream time slots for the connection between each PBX terminal and its corresponding NGN terminal based on the call connection information corresponding to each call connection; and performing voice stream conversion and communication between each PBX terminal and its corresponding NGN terminal based on the allocated voice stream time slots. The present application creates a voice call connection for each PBX user sharing the same VoIP account and its corresponding NGN terminal user, and allocates voice stream time slots for them, so that each PBX user and its corresponding NGN terminal user realize voice stream conversion and communication based on the voice stream time slot. Therefore, the voice communications between different PBX users sharing the same VoIP account and their corresponding NGN terminal users are independent of each other, thereby realizing the successful access of multi-channel voice of the same account under the PBX. BRIEF DESCRIPTION OF THE DRAWINGS

[0058] In order to more clearly illustrate the technical solutions in the embodiments of the present application, the drawings required for use in the description of the embodiments will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present application. For ordinary technicians in this field, other drawings can be obtained based on these drawings without creative work.

[0059] Figure 1 A flowchart of a method for implementing multiple voice calls with the same VoIP account provided in an embodiment of the present application;

[0060] Figure 2 A schematic diagram of a structure of two users and two-way voices sharing a common account provided in an embodiment of the present application;

[0061] Figure 3 A schematic diagram of a process of making a simultaneous call when two users of a common account make the call in an embodiment of the present application;

[0062] Figure 4 A schematic diagram of a process for simultaneously calling two users of a common account when they are called is provided in an embodiment of the present application. DETAILED DESCRIPTION

[0063] In order to make the purpose, technical solution and advantages of the embodiments of the present application clearer, the technical solution in the embodiments of the present application will be clearly and completely described below in conjunction with the drawings in the embodiments of the present application. Obviously, the described embodiments are part of the embodiments of the present application, not all of the embodiments. Based on the embodiments in the present application, all other embodiments obtained by ordinary technicians in this field without making creative work are within the scope of protection of this application.

[0064] The embodiment of the present application provides a method and system for implementing multi-channel voice calls based on the same account, which can solve the problem in the related art that when the PBX is integrated into the NGN end, the access of multi-channel voices under the same account under the PBX cannot be implemented.

[0065] Figure 1 A method for implementing multiple voice calls with the same VoIP account provided in an embodiment of the present application includes the following steps:

[0066] Step S10: establishing a call connection between multiple PBX terminals using the same VoIP account and corresponding NGN terminals;

[0067] Further, when the PBX end is used as the calling end and the NGN end is used as the called end, step S10 specifically includes the following steps:

[0068] The ISDN control module receives and parses the ISDN SETUP signaling (an ISDN communication protocol) sent by the first PBX end, obtains the call information of the first ISDN voice, the call information includes the ISDN calling number, the VoIP called number and the ISDN time slot, and creates a first ISDN call unit for processing the first ISDN voice according to the ISDN calling number and the ISDN time slot;

[0069] The first ISDN calling unit converts the ISDN calling number into a VoIP calling number, and sends the VoIP calling number, the VoIP called number and the ISDN time slot to the VoIP protocol module;

[0070] The VoIP protocol module creates a first VoIP signaling unit for processing a first ISDN voice according to the VoIP calling number and the ISDN time slot;

[0071] The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP calling number, sends the VoIP calling number, the VoIP called number and the calling end SDP (Session Description Protocol) message to the NGN end corresponding to the first PBX end, and obtains the response signaling sent by the NGN end according to the VoIP calling number, the VoIP called number and the calling end SDP message, wherein the response signaling includes the called end SDP message;

[0072] The first VoIP signaling unit converts the response signaling into a call CONNECT message, and sends the call CONNECT message to the ISDN control module;

[0073] The first ISDN call unit in the ISDN control module constructs ISDN CONNECT signaling (an ISDN communication protocol) according to the call CONNECT message, and sends the ISDN CONNECT signaling to the first PBX end, so that the first PBX end establishes a call connection with its corresponding NGN end based on the ISDN CONNECT signaling.

[0074] Further, when the PBX end serves as the called end and the NGN end serves as the calling end, step S10 specifically includes the following steps:

[0075] The VoIP protocol module receives and parses the SIP INVITE signaling sent by the first NGN end, obtains the call information of the first ISDN voice, the call information includes the VoIP called number, the calling end SDP message and the signaling key value, and creates a first VoIP signaling unit for processing the first ISDN voice according to the VoIP called number and the signaling key value;

[0076] The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP called number, and sends the VoIP called number to the ISDN control module;

[0077] The ISDN control module converts the VoIP called number into an ISDN called number, allocates an ISDN time slot to the first ISDN voice according to the ISDN called number, and returns the ISDN time slot to the first VoIP signaling unit;

[0078] The first VoIP signaling unit converts the SIP INVITE signaling into a call SETUP message, and sends the call SETUP message to the ISDN control module.

[0079] The ISDN control module creates a first ISDN call unit for processing the first ISDN voice according to the ISDN called number and the ISDN time slot;

[0080] The first ISDN call unit constructs ISDN SETUP signaling (an ISDN communication protocol) according to the call SETUP message, sends the ISDN SETUP signaling to the PBX end corresponding to the first NGN end, and obtains ISDN CONNECT signaling (an ISDN communication protocol) sent by the PBX end according to the ISDN SETUP signaling;

[0081] The first ISDN call unit converts the ISDN CONNECT signaling into a call CONNECT message, and sends the call CONNECT message to the VoIP protocol module;

[0082] The first VoIP signaling unit in the VoIP protocol module constructs a response signaling according to the call CONNECT message and sends it to the first NGN end. The response signaling includes the called end SDP message, so that the first NGN end can establish a call connection with its corresponding PBX end based on the response signaling.

[0083] Step S20: allocating a voice stream time slot for the connection between each PBX end and its corresponding NGN end based on the call connection information corresponding to each call connection;

[0084] Furthermore, step S20 specifically includes the following steps:

[0085] The ISDN PRI access module receives the call connection information of the first ISDN voice sent by the first VoIP signaling unit, wherein the call connection information includes the PRI port, the ISDN time slot, the calling end SDP message and the called end SDP message;

[0086] The ISDN PRI access module creates a first voice stream unit for processing a first ISDN voice according to the PRI port and the ISDN time slot;

[0087] The first voice stream unit configures parameters of a certain idle DSP time slot according to the SDP message of the calling end and the SDP message of the called end to obtain the first idle DSP time slot, and connects the first idle DSP time slot with the ISDN time slot based on a preset time slot mapping relationship to form a voice stream time slot between the PBX end and its corresponding NGN end.

[0088] Step S30: Based on the allocated voice stream time slot, voice stream conversion and communication are performed between each PBX end and its corresponding NGN end.

[0089] Furthermore, step S30 specifically includes the following steps:

[0090] The first voice stream unit converts the PCM voice stream of the PBX end into a first VoIP voice stream based on the SDP message of the called end, and the first VoIP voice stream is transmitted to the NGN end through the voice stream time slot;

[0091] The first voice stream unit converts the VoIP voice stream of the NGN end into a first PCM (Pulse Code Modulation) voice stream based on the SDP message of the calling end, and the first PCM voice stream is transmitted to the PBX end through the voice stream time slot.

[0092] For example, in this embodiment, since the same PBX external line account shared by multiple PBX users corresponds to the same VoIP account on the VoIP gateway, when the PBX is integrated into the NGN end, there is a problem of simultaneous access of multiple voices by the same PBX external line account under the PBX. Therefore, in order to enable multiple PBX users to use the same VoIP account to implement simultaneous calls of multiple voices on the NGN end, it is necessary to implement a key technology for converting concurrent multiple ISDN voices into concurrent VoIP multiple voices and accessing them to the NGN end. Among them, in order to better understand the specific technical solutions disclosed in the embodiments of the present application, the following will explain the multiple accounts involved in the specific embodiments: ISDN external line account refers to the ISDN protocol, which corresponds to the calling number field or the called number field in the ISDN protocol; PBX external line account refers to the external line account for the PBX protocol; VoIP account refers to the number configured for the VoIP gateway, which will be registered to the NGN network for use in calls on the NGN network; in addition, when the PBX external line account is connected to the NGN network, it cannot be directly used to access the NGN network using the PBX external line account, but the PBX external line account needs to be converted into a VoIP account before it can be used in the NGN network; VoIP calling number refers to the VoIP account as the calling party, and VoIP called number refers to the VoIP account as the called party; ISDN calling number corresponds to the calling number field in the ISDN protocol, that is, the PBX external line account as the calling party; ISDN called number corresponds to the called number field in the ISDN protocol, that is, the PBX external line account as the called party.

[0093] See also Figure 2As shown, in order to solve the above problems, this embodiment sets up a VoIP protocol module, an ISDN control module and an ISDN PRI access module, and creates a corresponding number of voice processing units for the same VoIP account according to the actual number of voice channels. One voice processing unit is used to access and process one ISDN voice channel. Each voice processing unit includes a VoIP signaling unit, an ISDN call unit and a voice flow unit to realize voice communication between multiple PBX end users and NGN end users who share the same account. Figure 3 Further explain the working principle of voice communication between multiple PBX-side users and NGN-side users who share the same account.

[0094] Since NGN signaling has different types such as Megaco and SIP, and the corresponding protocol commands are different, this application takes SIP as an example for explanation; and takes two users (i.e. Phone1 and Phone2) with a common account, with the PBX end as the calling end and the NGN end as the called end, as an example of the voice communication of connecting two ISDN voice channels to the NGN end:

[0095] Phone1 and Phone2 on the PBX side pick up the handset and dial at the same time. Since the voice control and exchange process of Phone1 and Phone2 are exactly the same, for the sake of brevity, the following only describes the workflow of Phone1: At this time, Phone1 will send ISDN SETUP signaling, which is transparently transmitted to the ISDN control module through the ISDN PRI access module based on the ISDN protocol;

[0096] The ISDN control module parses the ISDN SETUP signaling and obtains the ISDN calling number, the VoIP called number and the ISDN time slot; then the ISDN calling number and the ISDN time slot are used as unit identifiers to search whether an ISDN call unit corresponding to the ISDN calling number and the ISDN time slot already exists in the ISDN control module. If not found, a new ISDN call unit is created; if found, the found ISDN call unit is released and a new ISDN call unit is created. The new ISDN call unit then modifies the ISDN call state and the call establishment control information of the unit according to the ISDN calling number, the VoIP called number, the ISDN time slot and the message type, wherein the control information includes but is not limited to the information unit, the message type, the call identifier and the like; the ISDN calling number is converted into the VoIP calling number through the number conversion rule, for example, the ISDN calling number is 123456 and the VoIP calling number is 80123456, then the replacement rule is to add the prefix 80 to the ISDN calling number when the ISDN calling number is converted into the VOIP calling number; The SETUP signaling is converted into a call SETUP message carrying parameters such as message type, VoIP calling number, VoIP called number, ISDN time slot, etc., and the call SETUP message is transmitted to the VoIP protocol module;

[0097] After receiving the call SETUP message transmitted by the ISDN call unit, the VoIP protocol module uses the VoIP caller number and ISDN time slot as the unit identifier to find out whether the VoIP signaling unit corresponding to the VoIP caller number and ISDN time slot already exists in the VoIP protocol module. If not found, a new VoIP signaling unit is created; if found, the found VoIP signaling unit is released and a new VoIP signaling unit is created. The new VoIP signaling unit then constructs a call object (wherein the call object includes the From, To, session ID and Contact fields required for constructing SIP signaling) and a transaction list according to the call SETUP message for sending SIP INVITE signaling (an NGN communication protocol). The SIP The INVITE signaling carries signaling key values ​​such as From-tag, To-tag, CallLeg-ID, as well as VoIP calling number, VoIP called number and calling end SDP message (the calling end SDP message contains the calling end voice stream IP information and the calling end UDP port information) and other information to identify the voice call; that is, the VoIP signaling unit sends the SIP INVITE signaling to the NGN end and modifies the call state of the VoIP signaling unit and the control information of the call establishment;

[0098] The VoIP protocol module obtains the response signaling (the response signaling is SIP200OK signaling) sent by the NGN end according to the SIP INVITE signaling, and parses the SIP 200OK signaling to extract the VoIP calling number, the VoIP called number, the signaling key value and the called end SDP message (the called end SDP message contains the called end voice stream IP information and the called end UDP port information); then finds the corresponding VoIP signaling unit according to the VoIP calling number and the signaling key value;

[0099] The VoIP signaling unit converts the SIP 200OK signaling into a call CONNECT message carrying parameters such as message type, VoIP calling number, VoIP called number and ISDN time slot, transmits the call CONNECT message to the ISDN control module, and modifies the call state of the VoIP signaling unit and the control information of call establishment;

[0100] The ISDN control module uses the VoIP calling number and ISDN time slot as identifiers to find the corresponding ISDN call unit. The ISDN call unit constructs ISDN CONNECT signaling according to the call CONNECT message, and transparently transmits the ISDN CONNECT signaling to the PBX through the ISDN PRI access module, and modifies the ISDN call state and call establishment control information of the ISDN call unit, so that a call connection is established between the PBX end and its corresponding NGN end;

[0101] The VoIP signaling unit searches for the corresponding PRI port through the VoIP calling number and based on the configuration relationship between the PRI port and the VoIP called number, and notifies the ISDN PRI access module of the PRI port, ISDN time slot, calling end SDP message and called end SDP message;

[0102] The ISDN PRI access module uses the PRI port and ISDN time slot as unit identifiers to search whether there is already a voice stream unit corresponding to the PRI port and ISDN time slot in the ISDN PRI access module. If not found, a new voice stream unit is created; if found, the found voice stream unit is released and a new voice stream unit is created. The new voice stream unit selects an idle DSP time slot, and configures the DSP parameters of the idle DSP time slot according to the calling end SDP message and the called end SDP message transmitted by the VoIP signaling unit; then the idle DSP time slot and the ISDN time slot are connected through the time slot channel through the preset time slot mapping relationship in the time slot mapping table, and the PCM voice stream of the PBX end is encoded into a VoIP voice stream according to the encoding type in the called end SDP message and sent to the NGN end. At the same time, the VoIP voice stream received from the NGN end is decoded into a PCM voice stream according to the encoding type in the calling end SDP message and sent to the PBX end, thereby realizing the voice stream interconnection between the NGN network and the ISDN network.

[0103] The following combination Figure 4 Taking two users (Phone 1 and Phone 2) with the same account, with the PBX as the called end and the NGN as the calling end, and connecting two ISDN voice channels to the NGN as an example, the working principle of voice communication between multiple PBX users and NGN users sharing the same account is further explained.

[0104] Since the voice control and exchange process of Phone1 and Phone2 are exactly the same, for the sake of brevity, the following only describes the workflow of Phone1:

[0105] After receiving the SIP INVITE signaling sent from the NGN end, the VoIP protocol module parses the SIP INVITE signaling and extracts the VoIP called number and signaling key values ​​such as From-tag, To-tag, CallLeg-ID, and the calling end SDP message (the calling end SDP message contains the calling end voice stream IP information and the calling end UDP port information) and other information to identify the voice call; then the VoIP called number and signaling key value are used as unit identifiers to find out whether the VoIP protocol module already has a VoIP signaling unit corresponding to the VoIP called number and signaling key value. If not found, a new VoIP signaling unit is created; if found, the found VoIP signaling unit is released and a new VoIP signaling unit is created. The new VoIP signaling unit then constructs a call object and a transaction list according to the SIP INVITE signaling for subsequent SIP signaling transmission; and the VoIP called number is transmitted to the ISDN control module to notify the ISDN control module to allocate an ISDN time slot for the voice;

[0106] The ISDN control module converts the VoIP called number into an ISDN called number through the number conversion rule, searches for and returns the idle ISDN time slot in the ISDN called number to the VoIP signaling unit; the VoIP signaling unit converts the SIP INVITE signaling into a call SETUP message (the call SETUP message contains parameters such as the VoIP calling number, the VoIP called number, and the ISDN time slot), and transmits the call SETUP message to the ISDN control module;

[0107] The ISDN control module converts the VoIP called number into an ISDN called number again, and uses the ISDN called number and ISDN time slot as unit identifiers to find out whether an ISDN call unit corresponding to the ISDN called number and ISDN time slot already exists in the ISDN control module. If not found, a new ISDN call unit is created; if found, the found ISDN call unit is released and a new ISDN call unit is created; the new ISDN call unit then constructs an ISDN SETUP signaling according to the ISDN called number and ISDN time slot in the call SETUP message, and sends the ISDN SETUP signaling to the PBX end through the ISDN PRI access module, and modifies the ISDN call state and call establishment control information of the ISDN call unit. At this time, the Phone1 user on the PBX end will ring;

[0108] After Phone1 on the PBX goes off-hook, the ISDN control module receives the ISDN CONNECT signaling sent by the PBX according to the ISDN SETUP signaling, parses the ISDN CONNECT signaling, extracts the ISDN called number and ISDN time slot, and uses the ISDN called number and ISDN time slot to find the corresponding ISDN call unit;

[0109] After the ISDN calling unit converts the ISDN called number into the VoIP called number, it converts the ISDN CONNECT signaling into a call CONNECT message (the call CONNECT message includes parameters such as message type, VoIP calling number, VoIP called number, ISDN time slot, etc.), transmits the call CONNECT message to the VoIP protocol module, and modifies the ISDN call state and call establishment control information of the ISDN calling unit;

[0110] The VoIP protocol module finds the corresponding VoIP signaling unit through the VoIP called number and ISDN time slot; the VoIP signaling unit constructs a response signaling (the response signaling is SIP 200OK signaling) carrying the called party SDP message (the called party SDP message contains the called party voice stream IP information and the called party UDP port information) according to the call CONNECT message, and sends the SIP 200OK signaling to the NGN end through the call object and transaction list, and modifies the call state and call establishment control information of the VoIP signaling unit, and at the same time sends the PRI port, ISDN time slot, called party SDP message and calling party SDP message bound to the VoIP called number to the ISDN PRI access module;

[0111] The ISDN PRI access module uses the PRI port and ISDN time slot transmitted by the VoIP signaling unit as unit identifiers, and searches for a voice stream unit corresponding to the PRI port and ISDN time slot in the ISDN PRI access module. If not found, a new voice stream unit is created; if found, the found voice stream unit is released and a new voice stream unit is created. The new voice stream unit then selects an idle DSP time slot, and configures DSP parameters for the idle DSP time slot according to the called end SDP message and the calling end SDP message transmitted by the VoIP signaling unit. Then, the idle DSP time slot and ISDN time slot are connected through a time slot channel through a time slot mapping relationship preset in a time slot mapping table. The PCM voice stream on the PBX side is encoded into a VoIP voice stream according to the encoding type in the calling end SDP message and sent to the NGN side. At the same time, the VoIP voice stream received from the NGN network is decoded into a PCM voice stream according to the encoding type in the called end SDP message and sent to the PBX side, thereby realizing voice stream intercommunication between the NGN network and the ISDN network.

[0112] It can be seen that the embodiment of the present application simultaneously creates a voice call connection for each PBX user sharing the same VoIP account and its corresponding NGN end user, and allocates voice stream time slots to them respectively, so that each PBX user and its corresponding NGN end user can realize voice stream conversion and communication based on the voice stream time slot. Therefore, the voice communications between different PBX users sharing the same VoIP account and their corresponding NGN end users are independent of each other, thereby realizing the successful access of multiple voices of the same account under the PBX.

[0113] An embodiment of the present application also provides a system for implementing multiple voice calls based on the same account, including a VoIP gateway, wherein the VoIP gateway is used to establish call connections between multiple PBX ends using the same VoIP account and corresponding NGN ends; based on call connection information corresponding to each call connection, a voice stream time slot is allocated for the connection between each PBX end and its corresponding NGN end; based on the allocated voice stream time slot, voice stream conversion and communication are performed between each PBX end and its corresponding NGN end.

[0114] Further, when the PBX end is used as the calling end and the NGN end is used as the called end, the VoIP gateway includes an ISDN control module and a VoIP protocol module;

[0115] The ISDN control module receives and parses the ISDN SETUP signaling sent by the first PBX terminal, obtains the call information of the first ISDN voice, the call information includes the ISDN calling number, the VoIP called number and the ISDN time slot, and creates a first ISDN call unit for processing the first ISDN voice according to the ISDN calling number and the ISDN time slot;

[0116] The first ISDN calling unit converts the ISDN calling number into a VoIP calling number, and sends the VoIP calling number, the VoIP called number and the ISDN time slot to the VoIP protocol module;

[0117] The VoIP protocol module creates a first VoIP signaling unit for processing a first ISDN voice according to the VoIP calling number and the ISDN time slot;

[0118] The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP calling number, sends the VoIP calling number, the VoIP called number and the calling end SDP message to the NGN end corresponding to the first PBX end, and obtains the response signaling sent by the NGN end according to the VoIP calling number, the VoIP called number and the calling end SDP message, wherein the response signaling includes the called end SDP message;

[0119] The first VoIP signaling unit converts the response signaling into a call CONNECT message, and sends the call CONNECT message to the ISDN control module;

[0120] The first ISDN call unit in the ISDN control module constructs ISDN CONNECT signaling according to the call CONNECT message, and sends the ISDN CONNECT signaling to the first PBX end, so that the first PBX end establishes a call connection with its corresponding NGN end based on the ISDN CONNECT signaling.

[0121] Further, when the PBX end serves as the called end and the NGN end serves as the calling end, the VoIP gateway includes an ISDN control module and a VoIP protocol module;

[0122] The VoIP protocol module receives and parses the SIP INVITE signaling sent by the first NGN end, obtains the call information of the first ISDN voice, the call information includes the VoIP called number, the calling end SDP message and the signaling key value, and creates a first VoIP signaling unit for processing the first ISDN voice according to the VoIP called number and the signaling key value;

[0123] The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP called number, and sends the VoIP called number to the ISDN control module;

[0124] The ISDN control module converts the VoIP called number into an ISDN called number, allocates an ISDN time slot to the first ISDN voice according to the ISDN called number, and returns the ISDN time slot to the first VoIP signaling unit;

[0125] The first VoIP signaling unit converts the SIP INVITE signaling into a call SETUP message, and sends the call SETUP message to the ISDN control module;

[0126] The ISDN control module creates a first ISDN call unit for processing the first ISDN voice according to the ISDN called number and the ISDN time slot;

[0127] The first ISDN call unit constructs an ISDN SETUP signaling according to the call SETUP message, sends the ISDN SETUP signaling to the PBX end corresponding to the first NGN end, and obtains the ISDN CONNECT signaling sent by the PBX end according to the ISDN SETUP signaling;

[0128] The first ISDN call unit converts the ISDN CONNECT signaling into a call CONNECT message, and sends the call CONNECT message to the VoIP protocol module;

[0129] The first VoIP signaling unit in the VoIP protocol module constructs a response signaling according to the call CONNECT message and sends it to the first NGN end. The response signaling includes the called end SDP message, so that the first NGN end can establish a call connection with its corresponding PBX end based on the response signaling.

[0130] Furthermore, the VoIP gateway further includes an ISDN PRI access module, and the ISDN PRI access module is specifically used for:

[0131] Receiving call connection information of a first ISDN voice sent by a first VoIP signaling unit, the call connection information including a PRI port, an ISDN time slot, a calling end SDP message, and a called end SDP message;

[0132] Creating a first voice stream unit for processing a first ISDN voice according to the PRI port and the ISDN time slot;

[0133] The first voice stream unit configures parameters of a certain idle DSP time slot according to the SDP message of the calling end and the SDP message of the called end to obtain the first idle DSP time slot, and connects the first idle DSP time slot with the ISDN time slot based on a preset time slot mapping relationship to form a voice stream time slot between the PBX end and its corresponding NGN end.

[0134] Furthermore, the first voice stream unit is further specifically used for:

[0135] Converting the PCM voice stream of the PBX end into a first VoIP voice stream based on the SDP message of the called end, wherein the first VoIP voice stream is transmitted to the NGN end through the voice stream time slot;

[0136] The VoIP voice stream of the NGN end is converted into a first PCM voice stream based on the SDP message of the calling end, and the first PCM voice stream is transmitted to the PBX end through the voice stream time slot.

[0137] For example, see Figure 2 As shown, in this embodiment, two users with a common account on the PBX side perform voice communication with a user on the NGN side as an example: Figure 2 The interactive relationship between the VoIP gateway, NGN server and PBX is described in the specification. The VoIP gateway includes three main modules: VoIP protocol module, ISDN control module and ISDN PRI access module. The VoIP protocol module interacts with the NGN server through the VoIP protocol, the ISDN control module interacts with the PBX through the ISDN protocol, and the ISDN PRI access module is connected to the PBX through the E1 physical line.

[0138] In this embodiment, multiple voice processing units will be created for the same VoIP account according to the actual number of voice channels, that is, one voice processing unit only accesses and processes one ISDN voice channel. Each voice processing unit includes a VoIP signaling unit, an ISDN call unit, and a voice stream unit, and the VoIP protocol module is responsible for identifying the VoIP signaling unit, the ISDN control module is responsible for identifying the ISDN call unit, and the ISDN PRI access module is responsible for identifying the voice stream unit. It can be seen that the same VoIP account has multiple VoIP signaling units, multiple ISDN call units, and multiple voice stream units for multi-user voice docking with the PBX account, thereby realizing the simultaneous conversion of multiple ISDN voice channels into multiple VoIP voice channels on the VoIP gateway. For example, the voice processing of Phone1 is performed through VoIP signaling unit 1, ISDN call unit 1, and voice stream unit 1; the voice processing of Phone2 is performed through VoIP signaling unit 2, ISDN call unit 2, and voice stream unit 2. Therefore, the multi-channel ISDN voice of the same account can be converted into VoIP voice through the voice processing unit and sent to the NGN network. At the same time, the multi-channel voice of the same account on the NGN network can also be converted into ISDN multi-channel voice through the voice processing unit and sent to the PBX user.

[0139] Specifically, the VoIP signaling unit connects with the NGN server through the VoIP account to achieve intercommunication at the voice signaling control level; the VoIP signaling unit is used to identify and obtain key information such as the VoIP account and signaling key values ​​such as From-tag, To-tag, CallLeg-ID, and ISDN time slot, PRI port, and uses the VoIP account, signaling key value, and ISDN time slot as the identification mark of the VoIP signaling unit. For example, the VoIP protocol module parses the SIP INVITE signaling received from the NGN server to obtain the VoIP account and signaling key value, or receives the VoIP account and ISDN time slot transmitted from the ISDN control module, and searches for an existing VoIP signaling processing unit or creates a new VoIP signaling processing unit based on the VoIP account and signaling key value or the VoIP account and ISDN time slot.

[0140] The VoIP signaling processing unit is also used to construct call objects and transaction lists for sending SIP signaling, and is responsible for the encapsulation and sending, receiving and parsing of the VoIP voice protocol, as well as the processing and status maintenance of calls. In addition, the VoIP account and call message containing ISDN time slot in the VoIP signaling processing unit are used to interact with the ISDN control module to achieve the conversion between the VoIP protocol and the ISDN protocol, and the PRI port and ISDN time slot in the VoIP signaling processing unit are used as the key identification of multiple voice channels under the same port of the ISDN PRI access module.

[0141] The ISDN call unit connects with the PBX through the ISDN protocol to achieve intercommunication of the ISDN protocol control layer. The ISDN protocol is terminated in this unit. The ISDN call unit is used to identify and obtain key information such as ISDN external line account, ISDN time slot and ISDN call status, and uses the ISDN external line account and ISDN time slot as the identification mark of the ISDN call unit. For example, the ISDN control module parses the ISDN protocol received from the PBX to obtain the ISDN external line account and ISDN time slot, and searches for existing ISDN call units or creates new ISDN call units based on the ISDN external line account and ISDN time slot. In addition, the ISDN call unit is mainly responsible for the conversion of PBX ISDN external line account and VoIP account, the allocation of ISDN time slot when ISDN users are called, and the encapsulation, sending, receiving and processing of the ISDN protocol.

[0142] The voice stream unit realizes the access of the voice stream of the VoIP gateway and PBX PRI user through the E1 interface. It is mainly used to obtain key information such as PRI port, ISDN time slot, DSP time slot, SDP message, and uses the PRI port and ISDN time slot as the identification mark of the voice stream unit. For example, the ISDN PRI access module finds an existing voice stream unit or creates a new voice stream unit based on the PRI port and ISDN time slot transmitted from the VoIP signaling unit. In addition, the voice stream unit is mainly used to determine the idle DSP time slot, and set the relevant voice stream parameters in the SDP message to the idle DSP time slot, and is used for the channel connection between the DSP time slot and the ISDN time slot, thereby realizing the conversion of the ISDN user PCM voice stream to the VoIP voice stream, and sending the VoIP voice stream to the NGN network; at the same time, the VoIP voice stream received from the NGN network is also converted into a PCM voice stream and sent to the PBX user.

[0143] In summary, the embodiment of the present application can realize the function of simultaneous multi-channel ISDN voice calls on the VoIP gateway by connecting the multi-channel voice processing unit of the same VoIP account with multiple PBX users. It can be seen that the embodiment of the present application can realize the function of simultaneous PBXISDN multi-channel voice calls on the NGN network through the same VoIP account without changing the existing network architecture and PBX business functions of the PBX, further improving the integration of ISDN and NGN networks, and solving the problem of multiple users sharing the same VoIP account in the PBX; at the same time, enterprises can choose to access the appropriate number of VoIP accounts according to their own needs, thereby reducing unnecessary external line accounts, reducing operating costs, and improving the utilization rate of external line numbers.

[0144] It should be noted that technical personnel in the relevant field can clearly understand that for the convenience and conciseness of description, the specific working process of the system and each module and unit described above can refer to the corresponding process in the aforementioned method embodiment for realizing multi-way voice calls based on the same account, and will not be repeated here.

[0145] It should be noted that, in this application, relational terms such as "first" and "second" are only used to distinguish one entity or operation from another entity or operation, and do not necessarily require or imply any such actual relationship or order between these entities or operations. Moreover, in this article, the terms "include", "comprise" or any other variants thereof are intended to cover non-exclusive inclusion, so that a process, method, article or system including a series of elements includes not only those elements, but also other elements not explicitly listed, or also includes elements inherent to such process, method, article or system. In the absence of further restrictions, an element defined by the sentence "comprises a ..." does not exclude the existence of other identical elements in the process, method, article or system including the element.

[0146] The above description is only a specific implementation of the present application, so that those skilled in the art can understand or implement the present application. Various modifications to these embodiments will be apparent to those skilled in the art, and the general principles defined herein can be implemented in other embodiments without departing from the spirit or scope of the present application. Therefore, the present application will not be limited to the embodiments shown herein, but will conform to the widest range consistent with the principles and novel features applied for herein.

Claims

1. A method for implementing multi-way voice calls based on the same account, characterized in that: The following steps are involved: Establish call connections between multiple PBXs using the same VoIP account and the corresponding NGN terminals; Allocate a voice stream time slot for the connection between each PBX end and its corresponding NGN end based on the call connection information corresponding to each call connection; Based on the allocated voice stream time slot, voice stream conversion and communication are performed between each PBX end and its corresponding NGN end; When the PBX terminal is used as the calling terminal and the NGN terminal is used as the called terminal, the step of establishing a call connection between multiple PBX terminals using the same VoIP account and corresponding NGN terminals includes: The ISDN control module receives and parses the ISDN SETUP signaling sent by the first PBX terminal, obtains the call information of the first ISDN voice, the call information includes the ISDN calling number, the VoIP called number and the ISDN time slot, and creates a first ISDN call unit for processing the first ISDN voice according to the ISDN calling number and the ISDN time slot; The first ISDN calling unit converts the ISDN calling number into a VoIP calling number, and sends the VoIP calling number, the VoIP called number and the ISDN time slot to the VoIP protocol module; The VoIP protocol module creates a first VoIP signaling unit for processing a first ISDN voice according to the VoIP calling number and the ISDN time slot; The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP calling number, sends the VoIP calling number, the VoIP called number and the calling end SDP message to the NGN end corresponding to the first PBX end, and obtains the response signaling sent by the NGN end according to the VoIP calling number, the VoIP called number and the calling end SDP message, wherein the response signaling includes the called end SDP message; The first VoIP signaling unit converts the response signaling into a call CONNECT message, and sends the call CONNECT message to the ISDN control module; The first ISDN call unit in the ISDN control module constructs ISDN CONNECT signaling according to the call CONNECT message, and sends the ISDN CONNECT signaling to the first PBX end, so that the first PBX end establishes a call connection with its corresponding NGN end based on the ISDN CONNECT signaling.

2. The method for implementing multiple voice calls based on the same account as claimed in claim 1, characterized in that: When the PBX terminal is used as the called terminal and the NGN terminal is used as the calling terminal, the step of establishing a call connection between multiple PBX terminals using the same VoIP account and corresponding NGN terminals includes: The VoIP protocol module receives and parses the SIP INVITE signaling sent by the first NGN end, obtains the call information of the first ISDN voice, the call information includes the VoIP called number, the calling end SDP message and the signaling key value, and creates a first VoIP signaling unit for processing the first ISDN voice according to the VoIP called number and the signaling key value; The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP called number, and sends the VoIP called number to the ISDN control module; The ISDN control module converts the VoIP called number into an ISDN called number, allocates an ISDN time slot to the first ISDN voice according to the ISDN called number, and returns the ISDN time slot to the first VoIP signaling unit; The first VoIP signaling unit converts the SIP INVITE signaling into a call SETUP message, and sends the call SETUP message to the ISDN control module; The ISDN control module creates a first ISDN call unit for processing the first ISDN voice according to the ISDN called number and the ISDN time slot; The first ISDN call unit constructs an ISDN SETUP signaling according to the call SETUP message, sends the ISDN SETUP signaling to the PBX end corresponding to the first NGN end, and obtains the ISDN CONNECT signaling sent by the PBX end according to the ISDN SETUP signaling; The first ISDN call unit converts the ISDN CONNECT signaling into a call CONNECT message, and sends the call CONNECT message to the VoIP protocol module; The first VoIP signaling unit in the VoIP protocol module constructs a response signaling according to the call CONNECT message and sends it to the first NGN end. The response signaling includes the called end SDP message, so that the first NGN end can establish a call connection with its corresponding PBX end based on the response signaling.

3. The method for implementing multi-way voice calls based on the same account according to claim 1 or 2, characterized in that: The method of allocating a voice stream time slot for a connection between each PBX end and its corresponding NGN end based on call connection information corresponding to each call connection includes: The ISDN PRI access module receives the call connection information of the first ISDN voice sent by the first VoIP signaling unit, wherein the call connection information includes the PRI port, the ISDN time slot, the calling end SDP message and the called end SDP message; The ISDN PRI access module creates a first voice stream unit for processing a first ISDN voice according to the PRI port and the ISDN time slot; The first voice stream unit configures parameters of a certain idle DSP time slot according to the SDP message of the calling end and the SDP message of the called end to obtain the first idle DSP time slot, and connects the first idle DSP time slot with the ISDN time slot based on a preset time slot mapping relationship to form a voice stream time slot between the PBX end and its corresponding NGN end.

4. The method for implementing multiple voice calls based on the same account as claimed in claim 3, characterized in that: The voice stream conversion and communication between the PBX calling terminal and its corresponding NGN called terminal based on the allocated voice stream time slot includes: The first voice stream unit converts the PCM voice stream of the PBX end into a first VoIP voice stream based on the SDP message of the called end, and the first VoIP voice stream is transmitted to the NGN end through the voice stream time slot; The first voice stream unit converts the VoIP voice stream of the NGN end into a first PCM voice stream based on the SDP message of the calling end, and the first PCM voice stream is transmitted to the PBX end through the voice stream time slot.

5. A system for implementing multi-way voice calls based on the same account, characterized in that: The VoIP gateway is used to establish call connections between multiple PBX terminals using the same VoIP account and corresponding NGN terminals; allocate a voice stream time slot for the connection between each PBX terminal and its corresponding NGN terminal based on call connection information corresponding to each call connection; and perform voice stream conversion and communication between each PBX terminal and its corresponding NGN terminal based on the allocated voice stream time slot; Wherein, when the PBX end is used as the calling end and the NGN end is used as the called end, the VoIP gateway includes an ISDN control module and a VoIP protocol module; The ISDN control module receives and parses the ISDN SETUP signaling sent by the first PBX terminal, obtains the call information of the first ISDN voice, the call information includes the ISDN calling number, the VoIP called number and the ISDN time slot, and creates a first ISDN call unit for processing the first ISDN voice according to the ISDN calling number and the ISDN time slot; The first ISDN calling unit converts the ISDN calling number into a VoIP calling number, and sends the VoIP calling number, the VoIP called number and the ISDN time slot to the VoIP protocol module; The VoIP protocol module creates a first VoIP signaling unit for processing a first ISDN voice according to the VoIP calling number and the ISDN time slot; The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP calling number, sends the VoIP calling number, the VoIP called number and the calling end SDP message to the NGN end corresponding to the first PBX end, and obtains the response signaling sent by the NGN end according to the VoIP calling number, the VoIP called number and the calling end SDP message, wherein the response signaling includes the called end SDP message; The first VoIP signaling unit converts the response signaling into a call CONNECT message, and sends the call CONNECT message to the ISDN control module; The first ISDN call unit in the ISDN control module constructs ISDN CONNECT signaling according to the call CONNECT message, and sends the ISDN CONNECT signaling to the first PBX end, so that the first PBX end establishes a call connection with its corresponding NGN end based on the ISDN CONNECT signaling.

6. The system for implementing multiple voice calls based on the same account as claimed in claim 5, characterized in that: When the PBX end is used as the called end and the NGN end is used as the calling end, the VoIP gateway includes an ISDN control module and a VoIP protocol module; The VoIP protocol module receives and parses the SIP INVITE signaling sent by the first NGN end, obtains the call information of the first ISDN voice, the call information includes the VoIP called number, the calling end SDP message and the signaling key value, and creates a first VoIP signaling unit for processing the first ISDN voice according to the VoIP called number and the signaling key value; The first VoIP signaling unit determines the PRI port of the first ISDN voice according to the VoIP called number, and sends the VoIP called number to the ISDN control module; The ISDN control module converts the VoIP called number into an ISDN called number, allocates an ISDN time slot to the first ISDN voice according to the ISDN called number, and returns the ISDN time slot to the first VoIP signaling unit; The first VoIP signaling unit converts the SIP INVITE signaling into a call SETUP message, and sends the call SETUP message to the ISDN control module; The ISDN control module creates a first ISDN call unit for processing the first ISDN voice according to the ISDN called number and the ISDN time slot; The first ISDN call unit constructs an ISDN SETUP signaling according to the call SETUP message, sends the ISDN SETUP signaling to the PBX end corresponding to the first NGN end, and obtains the ISDN CONNECT signaling sent by the PBX end according to the ISDN SETUP signaling; The first ISDN call unit converts the ISDN CONNECT signaling into a call CONNECT message, and sends the call CONNECT message to the VoIP protocol module; The first VoIP signaling unit in the VoIP protocol module constructs a response signaling according to the call CONNECT message and sends it to the first NGN end. The response signaling includes the called end SDP message, so that the first NGN end can establish a call connection with its corresponding PBX end based on the response signaling.

7. The system for implementing multiple voice calls based on the same account as claimed in claim 5 or 6, characterized in that: The VoIP gateway further includes an ISDN PRI access module, and the ISDN PRI access module is specifically used for: Receiving call connection information of a first ISDN voice sent by a first VoIP signaling unit, the call connection information including a PRI port, an ISDN time slot, a calling end SDP message, and a called end SDP message; Creating a first voice stream unit for processing a first ISDN voice according to the PRI port and the ISDN time slot; The first voice stream unit configures parameters of a certain idle DSP time slot according to the SDP message of the calling end and the SDP message of the called end to obtain the first idle DSP time slot, and connects the first idle DSP time slot with the ISDN time slot based on a preset time slot mapping relationship to form a voice stream time slot between the PBX end and its corresponding NGN end.

8. The system for implementing multiple voice calls based on the same account as claimed in claim 7, characterized in that: The first voice stream unit is further specifically used for: Converting the PCM voice stream of the PBX end into a first VoIP voice stream based on the SDP message of the called end, wherein the first VoIP voice stream is transmitted to the NGN end through the voice stream time slot; The VoIP voice stream of the NGN end is converted into a first PCM voice stream based on the SDP message of the calling end, and the first PCM voice stream is transmitted to the first PBX end through the voice stream time slot.

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