CN1874380A - Method for realizing a call - Google Patents

Abstract

The invention provides a method for implementing a call. After the first terminal establishes communication with the second terminal using the SIP protocol, the first softswitch device suspends the first terminal after receiving the SIP signaling triggered by the hook flashing of the first terminal. And notify the second softswitch equipment to suspend the second terminal by SIP signaling; after the first softswitch equipment collects the number of the third terminal sent by the first terminal, establish the connection to the third terminal by SIP signaling The connection of the 3rd softswitch equipment, and the service description protocol SDP of multi-party conversation is sent to the 3rd terminal that the 3rd softswitch equipment is connected in the process of establishing connection; Then, the first softswitch equipment sends the business of described multiparty conversation The description protocol SDP is sent to the first terminal; the first and third terminals establish a connection according to the service description protocol SDP of the multi-party call to conduct a call between them. The invention realizes flexible conversation under the SIP protocol, and can be used for multi-party conversation.

Description

How to implement the call

technical field

The invention relates to the technical field of SIP, in particular to a method for implementing a call.

Background technique

PSTN network is evolving to NGN network, and operators are gradually turning to NGN-based network solutions. Because it is easy to realize new value-added services and its openness and distribution, NGN has gradually become the new darling of many operators and has been listed as their investment plan. Providing various supplementary services (Supplementary Services) for NGN users has become an urgent task of NGN network solutions. NGN is a unified general network solution, which includes IP-based users and network protocols (such as SIP, H.323, etc.). Among them, the SIP protocol has become one of the most widely used protocols due to its openness and simplicity.

In the process of evolution to NGN, NGN must be able to provide users with all supplementary services similar to those provided by PSTN, and for end users, their way of using supplementary services should remain the same as that used in PSTN networks Same, it is convenient for users to accept supplementary services in NGN. At present, in the NGN network, the implementation methods for supplementary services can be mainly classified into the following two solutions: terminal control mode and network control mode.

The terminal control mode is a new implementation mode, which requires the terminal itself to be more intelligent, realize more business logic and have stronger business control capabilities. At present, there is still a lack of relevant technical standards, and the required intercommunication capabilities between terminals and other network devices are not satisfactory. Therefore, this method leads to the following defects: the operation mode of the user is inconsistent with the operation mode of the PSTN user; it is difficult to manage the service availability of the user because the service is triggered at the terminal; the call process requires close cooperation between the terminal and the network server, and any new The implementation of the service requires the upgrade support of the terminal and the network server; before triggering the service, the terminal must decide what service can be triggered according to the user's service attributes and call context details.

The network control method is an implementation scheme that adopts a centralized control method. Different network protocols can be used between the terminal (such as SIP IAD) and the softswitch device, but it is completely transparent to the user (that is, there is no need to change the user's usage), the user can still use the POTS phone. However, there is no supplementary service implementation standard or specification based on the network control method at present.

In PSTN, the multi-party call service is a practical supplementary service, and its derived services also include three-party calls, conference calls, and so on. It is an inevitable trend to use SIP protocol in NGN to realize multi-party calls, but as mentioned above, most of the current supplementary services in NGN are implemented in terminal control mode, which has the above disadvantages; and for using network control mode, there is no Realize relevant standards for supplementary services such as multi-party calls and conference TV.

Contents of the invention

In view of this, the main purpose of the present invention is to provide a call implementation method to realize the flexible call based on the network control mode under the SIP protocol, and then can be used to realize supplementary services such as multi-party calls.

The present invention provides a method for implementing a call. After the first terminal uses the SIP protocol to establish communication with the second terminal, it includes the following steps:

A1. After the first softswitch connected to the first terminal receives the SIP signaling triggered by the first terminal’s flashing, it suspends the first terminal, and notifies the second softswitch connected to the second terminal to hang up through SIP signaling. start the second terminal;

After B1, the first softswitch device collects the number of the third terminal sent by the first terminal, it establishes a connection to the third softswitch device connected to the third terminal through SIP signaling, and in the process of establishing a connection, multiple parties The service description protocol SDP of the call is sent to the third terminal connected to the third softswitch device;

C1. The first softswitch device sends the service description protocol SDP of the multi-party call to the first terminal;

D1. The first and third terminals establish a connection according to the service description protocol SDP of the multi-party call to conduct a call between them.

Wherein, step C1 further includes: the first softswitch device sends the service description protocol SDP of the multi-party call to the second terminal through the second softswitch device;

Step D1 further includes: the second terminal establishes a connection with the first terminal and the third terminal according to the service description protocol SDP of the multi-party call to carry out the multi-party call.

Wherein, after step D1, further include:

A3. After the first softswitch device receives the SIP signaling triggered by the hook flashing from the first terminal, it suspends the first terminal, and notifies the third softswitch device to suspend the third terminal through the SIP signaling;

B3. After receiving the SIP signaling triggered by the specific key of the first terminal, the first softswitch sends the service description protocol SDP of the multi-party call to the second terminal through the second softswitch;

C3. The first terminal and the second terminal establish a connection according to the service description protocol SDP of the multi-party call to carry out the call between them.

Wherein, after step D1, further include:

A4. After the first softswitch device receives the SIP signaling triggered by the hook flashing from the first terminal, it suspends the first terminal, and notifies the third softswitch device to suspend the third terminal through the SIP signaling;

B4. After the first softswitch receives the SIP signaling triggered by the specific key of the first terminal, the second softswitch notifies the second softswitch of the release instruction to release the second terminal through the second softswitch;

C4. The first terminal and the third terminal establish a connection according to the service description protocol SDP of the multi-party call to carry out the call between them.

Wherein, after A1, A3 or A4, it further includes: the first softswitch requests the service tone resource, and then plays the service prompt tone to the first terminal; plays the suspension tone to the suspended terminal.

Wherein, the terminal is a SIP terminal, and the SIP terminal generates the specific SIP signaling message according to the cross-click of the user; or,

The terminal includes a POTS terminal and a SIP gateway. The POTS terminal transmits the cross-click signal of the user to the SIP gateway, and the SIP gateway generates the specific SIP signaling message after receiving it.

Wherein, the SIP gateway includes: a SIP access gateway SIP AG or a SIP integrated access device SIP-IAD.

Wherein, before step B1, it further includes: the first softswitch device applies for a conference resource for the multi-party call, and the SDP of the multi-party call is an SDP including the conference resource.

The present invention provides another method for implementing a call. After the first terminal uses the SIP protocol to establish communication with a certain terminal, it includes the following steps:

A9. After the first softswitch device receives the SIP signaling triggered by the hook flashing from the first terminal, it suspends the first terminal, and notifies other connected softswitch devices to suspend other terminals through the SIP signaling;

B9. After the first softswitch equipment collects the number of the next terminal sent by the first terminal, it establishes a connection to the softswitch equipment connected to the next terminal through SIP signaling, and in the process of establishing the connection, the multi-party conversation The service description protocol SDP is transmitted to the next terminal connected to the switching device;

C9. The first softswitch device sends the service description protocol SDP of the multi-party call to the suspended terminal;

D9. Establish the multiparty call between the above terminals according to the service description protocol SDP of the multiparty call.

As can be seen from the above method, the present invention provides flexible call services for SIP users based on the network control method, which can not only realize simultaneous calls of multiple parties, but also switch calls between different users, thereby realizing flexible multi-party calls and based on Conference call for multiple parties. Since the network control method is adopted, the business logic of the multi-party call is completed on the network server side (such as a soft switch device), and the distribution and management of network resources (such as conference resources, voice resources, etc.) are also provided by the network server side, and the network resources are provided by the network server. Shared use by multiple users also provides service providers with good service manageability and controllability. Moreover, since the network control method is adopted to realize the flexible calling service, the coupling between the SIP terminal and the SIP server is separated, and the implementing method of the calling service will not change the user's operation mode when using the service.

Description of drawings

Figure 1 is a flow chart of establishing a multi-party call.

Fig. 2 is a flow chart of establishing a multi-party call in which the third party hangs up the two-party call.

Fig. 3 is a flow chart of resuming a multi-party call for a suspended user.

Fig. 4 is a flow chart of releasing a suspended user in the process of establishing a multi-party call.

Detailed ways

This paper proposes a solution to provide flexible calls for SIP users in the NGN network, so as to realize flexible switching during user calls and calls between different users, as well as flexible multi-party calls. The present invention will be described in detail below with reference to the accompanying drawings and embodiments.

First, the present invention is described in detail with reference to the flow chart of establishing a multi-party call shown in Figure 1. In this example, the user of SIP terminal A has the authority to initiate and manage the multi-party call service, and calls SIP terminal B at SIP terminal A and enters the conversation state Afterwards, the user of SIP terminal A calls the user of SIP terminal C by flashing fork, and initiates a multi-party call between SIP terminals A, B, and C. The method for establishing a multi-party call includes the following steps:

Step 101: The user flashes the hook, and after the SIP terminal A detects the hook flash event, it triggers a re-INVITE (re-invitation) message and reports it to the softswitch A. The SDP (Session Description Protocol) of this re-INVITE is HOLD SDP.

Step 102: Softswitch A receives the re-INVITE, applies for a HOLD tone resource to an available resource server, and sends a re-INVITE message to SIP terminal B through softswitch B after the application is successful, and the SDP in the message is also a HOLD SDP , to suspend SIP terminal B. After the call of SIP terminal B is suspended, the original RTP media stream of the call will also be suspended; the SDP in this message also contains media resources, so that the user of SIP terminal B can listen to the HOLD sound.

Step 103: Softswitch A returns a response message (200/ACK) to SIP terminal A, and the response message instructs SIP terminal A to send a service tone to the user to prompt the user to dial.

Step 104: After receiving the response message, SIP terminal A sends a service tone to the user, and starts to collect numbers to collect the number of SIP terminal C dialed in by the user.

Step 105: After the number collection operation is completed, SIP terminal A reports the number of SIP terminal C collected to softswitch A through an INFO message.

Step 106: After obtaining the reported number of the SIP terminal C, the softswitch A initiates a call request to the SIP terminal C after judging that all the numbers have been received. The SDP carrying conference resources in the call request message initiated to the SIP terminal C may be called CONF SDP.

In order to prevent the failure of applying for conference resources after SIP terminal C is connected, and to directly enter the conference after SIP terminal C goes off-hook, you can apply for conference resources locally before initiating a call request to SIP terminal C. The call request message initiated by terminal C carries the SDP of the conference resource.

Step 107: After receiving the off-hook signal from the SIP terminal C, the softswitch A is responsible for initiating a media modification process to the SIP terminals A and B, and adding the SIP terminals A and B into the conference. The modification process is as follows: the softswitch A initiates a re-INVITE process to the SIP terminals A and B respectively, and the re-INVITE request message carries the SDP of the conference resource.

Afterwards, SIP terminals A and B receive the re-INVITE message, and join the current conference according to the SDP of the conference resource, realizing the multi-party call of SIP terminals A, B and C.

At any time, SIP terminal A receives the user's hook flash again and calls other users such as SIP terminal D. After SIP terminal D answers, he will also be added to the conference, and will not be described again.

As shown in Fig. 2, according to the actual business needs, after calling the SIP terminal C, the call between the SIP terminals A and C can be realized, and the current SIP terminal B is suspended. To realize this situation, it is necessary to not initiate the media modification process to SIP terminal B in step 107, but only initiate the media modification process to SIP terminal A, add SIP terminal A to the conference, and realize that SIP terminals A and C have entered the conference state to make a call, but SIP terminal B is still in the suspended state.

For the embodiment of FIG. 2, when SIP terminal A and C are talking, and SIP terminal B is in a suspended state, SIP terminal A can still flash the hook again to initiate a three-party call service. Referring to FIG. 3, it includes the following steps:

Step 301: The user flashes the hook again. After the SIP terminal A detects the hook flashing event, it triggers a re-INVITE (re-invitation) message and reports it to the softswitch A. The SDP (Session Description Protocol) of this re-INVITE is HOLD SDP.

Step 302: Softswitch A receives said re-INVITE, notifies Softswitch C that SIP terminal C is suspended, and Softswitch A returns a response message (200/ACK) to SIP terminal A, and this response message indicates that the SIP terminal sends a message to the user. Send business tone.

Step 303: SIP terminal A performs number collection operation, collects the numbers dialed in by the user, and reports to softswitch A through an INFO message. In this example, according to the service requirements, the number dialed in by the collected user is "1", indicating that the user enters into the three-party call state.

Step 304: After receiving the user's dialing selection, softswitch A allocates conference resources if they have not been allocated. In this example, since conference resources have been allocated, the media modification process is directly initiated to SIP terminals A, B, and C. The re-INVITE request message in the media modification process will carry the SDP of the conference resource.

Afterwards, the SIP terminals A, B, and C respond to access the conference system. Realize the three-way call between them.

For the embodiment of Fig. 3, when SIP terminal A and C are talking, when SIP terminal B is in the suspended state, in step 303, if according to service requirements, collect the number dialed in by the user to represent the release of the suspended When the number of SIP terminal B is "3", refer to Figure 4, the corresponding subsequent steps are: after receiving the user's dialing selection, softswitch A instructs softswitch B to release the resources of SIP terminal B, and at the same time instructs softswitch C to resume SIP A conference call between terminals A and C. This process is the process of flashing the hook to initiate the release of the suspended SIP terminal B.

For the embodiment of Fig. 3, when SIP terminal A and C are talking, when SIP terminal B is in the suspended state, in step 303, if according to business requirements, collect the number dialed in by the user to indicate switching to and suspending When the SIP terminal B communicates with the number "4", the corresponding subsequent steps are: after receiving the user's dial selection, softswitch A instructs softswitch B to resume the conference call with SIP terminal B; at the same time instructs softswitch C to Suspend SIP terminal C. This process is the process of flashing the mobile phone to initiate a handover with the suspended user B.

The SIP terminal mentioned above can be a combination of POTS phones and SIP gateways, and the SIP gateways can be SIP AG (SIP Access Gateway), SIP-IAD (SIP Integrated Access Device) and other devices. The SIP gateway is used to realize the access of the POTS phone, digitize the voice and package it into IP and transmit it to the SIP service control unit, and communicate with the POTS through telecommunication signaling, and use SIP to communicate with the SIP service control unit. The POTS accesses the network through the SIP gateway, and the flashing signal sent by the POTS is an analog signal. After being received by the SIP gateway, it triggers the generation of a re-INVITE message. Of course, for a SIP terminal that supports the SIP protocol, the re-INVITE trigger information can be directly triggered according to the hook flashing, which is also suitable for the present invention.

As can be seen from the above method, the present invention provides flexible call services for SIP users based on the network control method, which can not only realize simultaneous calls of multiple parties, but also switch calls between different users, thereby realizing flexible call control, and then It can provide users with multi-party calls and conference calls based on multi-party calls.

The above descriptions are only preferred embodiments of the present invention, and are not intended to limit the present invention. Any modifications, equivalent replacements, improvements, etc. made within the spirit and principles of the present invention shall be included in the scope of the present invention. within the scope of protection.

Claims (9)

1, a kind of implementation method of conversation is characterized in that, first terminal may further comprise the steps after using communicating by letter of the Session Initiation Protocol foundation and second terminal:

After the SIP signaling that first terminal of receiving A1, first Softswitch that connects first terminal is triggered by hooking, hang up first terminal, and hang up second terminal by second Softswitch that the SIP signaling connects second terminal;

After B1, first Softswitch are collected the number of the 3rd terminal that first terminal sent, be established to the connection of the 3rd Softswitch that the 3rd terminal connected by the SIP signaling, and in the process of connecting, send the business description agreement SDP of MPTY to the 3rd terminal that the 3rd Softswitch is connected;

C1, first Softswitch send to first terminal with the business description agreement SDP of described MPTY;

D1, first and third terminal are according to connect conversation between carrying out of the business description agreement SDP of described MPTY.

2, method according to claim 1 is characterized in that, step C1 further comprises: first Softswitch sends to second terminal by second Softswitch with the business description agreement SDP of described MPTY;

Step D1 further comprises: second terminal is set up to be connected with first and third terminal according to the business description agreement SDP of described MPTY and is carried out Three-Way Calling.

3, method according to claim 1 is characterized in that, further comprises behind the step D1:

A3, first Softswitch are hung up first terminal, and are hung up the 3rd terminal by SIP signaling the 3rd Softswitch after receiving the SIP signaling of first terminal by the hooking triggering;

After the SIP signaling that the first terminal specific keys of receiving B3, first Softswitch triggers, the business description agreement SDP of described MPTY is sent to second terminal by second Softswitch;

C3, first and second terminal are according to connect conversation between carrying out of the business description agreement SDP of described MPTY.

4, method according to claim 1 is characterized in that, further comprises behind the step D1:

A4, first Softswitch are hung up first terminal, and are hung up the 3rd terminal by SIP signaling the 3rd Softswitch after receiving the SIP signaling of first terminal by the hooking triggering;

After the SIP signaling that the first terminal specific keys of receiving B4, first Softswitch triggers, will release order by second Softswitch and to notify second Softswitch to discharge second terminal;

Conversation between C4, first and third terminal are carried out according to the business description agreement SDP recovery connection of the MPTY of storage.

5, method according to claim 1 is characterized in that, further comprises behind A1, A3 or the A4: the first Softswitch requested service sound resource, then to the first terminal plays service informing sound; Play the hang-up sound by the Softswitch that held terminal connected to held terminal.

6, method according to claim 1 is characterized in that, described terminal is a sip terminal, produces the SIP signaling that described hooking triggers according to user's hooking sip terminal; Or,

Described terminal comprises POTS terminal and SIP gateway, and the POTS terminal passes to the SIP gateway with described user's hooking signal, and the SIP gateway receives that the back produces the SIP signaling that described hooking triggers.

7, method according to claim 6 is characterized in that, described SIP gateway comprises:

SIP IAD SIP AG or SIP integrated access equipment SIP-IAD.

8, method according to claim 1 is characterized in that, further comprise before the step B1: the first Softswitch application is used for the conferencing resource of MPTY, and the business description agreement SDP of described MPTY is the SDP that comprises described conferencing resource.

9, a kind of implementation method of conversation is characterized in that, first terminal may further comprise the steps after using communicating by letter of Session Initiation Protocol foundation and certain terminal:

A9, first Softswitch are hung up first terminal after receiving the SIP signaling of first terminal by the hooking triggering, and hang up the other-end that is in the conversation of first Softswitch by other Softswitch of SIP signaling;

After B9, first Softswitch are collected the number of next terminal that first terminal sent, be established to the connection of next Softswitch that next terminal connects by the SIP signaling, and in the process of connecting, send the business description agreement SDP of MPTY to next terminal that described next Softswitch connects;

C9, first Softswitch send to institute's held terminal with the business description agreement SDP of described MPTY;

D9, set up MPTY between above-mentioned each terminal according to the business description agreement SDP of described MPTY.