CN114710821A

CN114710821A - Method, system and storage medium for VoLTE relay to access SIP contact center

Info

Publication number: CN114710821A
Application number: CN202210254975.1A
Authority: CN
Inventors: 彭勇; 张进财
Original assignee: Shanghai Jingxing Information Technology Co ltd
Current assignee: Shanghai Jingxing Information Technology Co ltd
Priority date: 2022-03-15
Filing date: 2022-03-15
Publication date: 2022-07-05

Abstract

The application relates to a method, a system and a storage medium for accessing a VoLTE relay to an SIP contact center, wherein the method comprises the steps of obtaining an IP address after load distribution based on receiving a first request from the VoLTE relay; generating a type mark corresponding to the type content based on preset type content analyzed from the first request, wherein the type content comprises video, audio or pictures; generating SIP signaling according to the IP address, the type mark and the first request based on the received streaming media corresponding to the first request; generating an SIP communication request according to the SIP signaling and the type content; sending the SIP communication request to an SIP contact center; the method and the system have the effect of enabling the traditional SIP contact center to achieve VoLTE voice and video call access.

Description

Method, system and storage medium for VoLTE relay to access SIP contact center

Technical Field

The present application relates to the field of communications, and in particular, to a method, a system, and a storage medium for accessing a VoLTE relay to an SIP contact center.

Background

VoLTE is Voice-over-Long-Term-Evolution, that is, a short for Long Term Evolution Voice bearer, is a high-speed wireless communication standard for mobile phones and data terminals, is a high-definition Voice and video call service based on 4G and 5G networks, and is also called a high-definition telephone service, including a high-definition Voice telephone and a high-definition video telephone. The most direct feeling brought to the user by the VoLTE technology is that the connection waiting time is shorter, the voice and video call effect is higher and more natural, along with the development of 4G and 5G networks, the speed of a mobile network is faster and faster, the bandwidth is larger and larger, therefore, more and more users open the VoLTE can directly use the mobile phone call function in the voice and video call of the VoLTE, the APP does not need to be installed, for a contact center, through the access of a VoLTE relay, the better call quality can be provided, and meanwhile, the video access capability which is not possessed by the traditional voice line can be provided.

However, the conventional SIP contact center can be supported in the internal communication system used by the existing bank, the latest VoLTE technology cannot be used yet, and if a set of internal communication system using the VoLTE technology is directly replaced, the cost is enormous, so that a method for enabling the conventional SIP contact center in the existing bank to access the VoLTE technology is required.

Disclosure of Invention

In order to enable the conventional SIP contact center in the existing bank to be capable of accessing the VoLTE technology, the application provides a method, a system and a storage medium for the VoLTE relay to access the SIP contact center.

In a first aspect, the present application provides a method for accessing a VoLTE relay to an SIP contact center, which adopts the following technical scheme:

a method for accessing a VoLTE relay to an SIP contact center comprises the following steps:

acquiring an IP address after loading based on receiving a first request from a VoLTE relay;

generating a type mark corresponding to the type content based on preset type content analyzed from the first request, wherein the type content comprises video, audio or pictures;

generating SIP signaling according to the IP address, the type mark and the first request based on the received streaming media corresponding to the first request;

generating an SIP communication request according to the SIP signaling and the type content;

and sending the SIP communication request to an SIP contact center.

By adopting the technical scheme, the IP address after the load is obtained according to the first request sent by the VoLTE relay, the preset type content is analyzed from the first request, the type mark is marked on the type content again, the subsequent SIP contact center can identify the type content in the streaming media, the SIP communication request generated according to the newly generated SIP signaling and the streaming media is sent to the SIP contact center, the SIP contact center in the intranet can identify the streaming media sent by the VoLTE relay through the SIP communication request, and the traditional SIP contact center can realize VoLTE voice and video communication.

Preferably, the step of receiving the first request from the VoLTE relay further includes:

monitoring an external network card;

after the proxy server of the external network card receives the first request, the NAT is identified;

and the media server of the external network card receives the streaming media.

By adopting the technical scheme, the external network card is monitored, and when a data packet containing the first request or the streaming media reaches the external network card, the NAT can be automatically identified, so that the IP address in the first request can be modified, a subsequent SIP contact center can acquire the correct IP address, and the media stream can reach the terminal corresponding to the correct IP address.

Preferably, the step of acquiring an IP address requiring load further includes the steps of:

after receiving the first request, acquiring resource use data on the current node;

calculating the maximum remaining available resource based on the resource usage data, and outputting an IP address corresponding to the maximum remaining available resource; and if the maximum remaining available resources are multiple, randomly outputting an IP address corresponding to one of the maximum remaining available resources.

By adopting the technical scheme, the IP address needing to be loaded can be quickly found, and the response speed of the internal network to the external network is improved.

Preferably, the step of generating a genre flag corresponding to the genre content based on parsing a preset genre content from the first request further includes:

extracting an invitation message in the first request;

identifying the type content of the media stream corresponding to the first request according to the content in the invitation message;

and matching a type mark from preset mark targets according to the type content.

By adopting the technical scheme, the content of the streaming media is marked according to the content in the invitation information, the streaming media does not need to be directly identified, and the processing speed of the algorithm is improved.

Preferably, the step before the first request from the VoLTE relay is received further includes:

acquiring an interactive node input by a calling party sending the first request;

analyzing a preset target instruction from the input information based on the input information of the calling party in the interactive node;

accessing the calling party and waiting for the first request or an invitation message from the first request according to the target instruction;

and connecting the calling party to the called party corresponding to the IP address based on the received invitation message.

By adopting the technical scheme, the calling party can select the video voice to enter the SIP contact center by dialing the incoming line number provided by the VoLTE relay, the target instruction can be the calling mode selected by the calling party by being guided, the called party corresponding to the calling mode can be selected technically, and the VoLTE relay is accessed in the intranet.

Preferably, the step of generating the SIP signaling according to the IP address, the type flag, and the first request further includes:

acquiring a codec list supported by a calling party which sends the first request;

matching and comparing the SDP in the SIP signaling with the parameters in the codec list;

negotiating out a coding mode matched with the calling party according to a coding sequence provided in the SDP;

and transcoding the streaming media based on the result of transcoding required after negotiation.

By adopting the technical scheme, transcoding operation is carried out in the process of generating the SIP signaling, if the calling party and the called party both support the same media format, transcoding is not needed, and if the calling party and the called party do not support the same media format, transcoding is needed.

Preferably, the method further comprises the steps of:

identifying the type tag based on a voice response interpretation process for the SIP communication request;

identifying the streaming media from the SIP communication request according to the type mark, and playing the streaming media according to a playing method corresponding to the type mark; wherein, the step of playing comprises playing video or playing pictures and audio.

By adopting the technical scheme, the video type can be divided into a pure video type and a picture type and a voice type by using the type mark, and the pure video can be directly identified and played, so that the effect of video IVR is realized; the picture + speech type is a composite mode, i.e. a video effect is achieved by a combination of a separate picture and a separate audio.

Preferably, the method further comprises the steps of:

before the step of playing the picture and the voice, setting a preset time length of the picture, a preset frequency of the audio and a preset track of a playing medium in the type mark;

and when the picture and the voice are played, displaying the picture to the preset time length and playing the audio frequency for the preset frequency according to a preset track.

By adopting the technical scheme, the time length of picture display and the frequency of audio playing can be set in the type mark, when the type mark is identified, the picture with fixed time length and the frequency of circularly playing the audio can be displayed, and finally the effect of dynamically allocating the combination of the picture and the audio is achieved.

In a second aspect, the present application provides a system for accessing a VoLTE relay to an SIP contact center, which adopts the following technical solution:

a system for VoLTE relay access to an SIP contact center runs a computer program of any one of the methods for VoLTE relay access to the SIP contact center.

In a third aspect, the present application provides a computer storage medium, which adopts the following technical solutions:

a computer storage medium storing a computer program that can be loaded by a processor and that performs any of the above methods for VoLTE relay access to a SIP contact center.

Drawings

Fig. 1 is a schematic overall method flow diagram of a method for accessing a VoLTE relay to an SIP contact center in the present application;

fig. 2 is a schematic diagram of a center architecture of a method for accessing a VoLTE relay to a SIP contact center in the present application;

fig. 3 is a schematic flowchart of a method for processing an INVITE message in a method for accessing a SIP contact center by a VoLTE relay in the present application.

Reference numerals: 1. a VoLTE relay; 2. SBC; 3. and a PBX.

Detailed Description

The present application is described in further detail below with reference to figures 1-3.

The embodiment of the application discloses a method for accessing a VoLTE relay to an SIP contact center. As shown in fig. 1 and fig. 2, a method for accessing a VoLTE relay to an SIP contact center relates to VoLTE relay 1, SBC2, PBX3 and seat application. VoLTE relay 1 is a relay point to access a VoLTE based primary caller whose terminal may be a smartphone or a landline. SBC2 is a VoIP access layer device that implements NAT or firewall penetration functionality by controlling sessions at the network boundaries, while also enabling bandwidth limiting, session management, traffic statistics, etc. SBC2 is also known as a proxy server supporting VoIP, which can recognize messages of the fifth and seventh layers, and can handle the multiple session signaling protocols above the fifth layer, and can change the address of the data header, thereby implementing the address conversion between the network and the intranet within SBC 2. The PBX3 is a user-level switch, i.e., a telephone service network used inside a company, and extension users inside the system share a certain number of external lines, and the switch may adopt a switching device in which an IVVR visual flow server, a registration server, a media server, and an application server are built. The IVVR visual process server provides an IVR interpreter based on vxml3.0 standard, and functions of visual navigation, audio and video broadcasting, key input and the like of video incoming calls can be realized through IVVR, so that a video IVR function can be realized; the registration server provides functions such as an extension registration function and a load balancing function; the media server realizes the functions of media negotiation transfer, call control, conversation media stream processing and the like; the application server comprises functions of CTI middleware, record report, routing management and the like.

The SIP contact center may include a PBX 3. The seat application is seat end related application which is mainly based on soft phone and WebRTC-SDK.

The method comprises the following steps:

SBC2 receives the first request from VoLTE relay 1. SBC2 is provided with an external network card and an internal network card, SBC2 includes a load balancing module and a plurality of media service modules, and the internal network card and the plurality of media service modules can perform data transmission. The load balancing module is mainly used as a proxy or a load server, and the media service module is mainly used as a media or business server. The load balancing module is an open source SIP proxy server for voice, video, IM, presence and any other SIP extensions, is a multifunctional, multi-purpose SIP signaling server that operators, telecom or ITSP use for Class4/5 platforms, relays, enterprise/virtual PBX3 solutions, session border controllers, application servers, front end load, IMs platforms, call centers, etc.

SBC2 supports two network cards to interface with the WAN and LAN. The external network card is connected with the WAN in a butt joint mode, the VoLTE relay 1 is connected with the WAN in a butt joint mode, the internal network card is connected with the LAN in a butt joint mode, and communication is carried out through the LAN and the PBX 3. The SBC2 monitors the external network card and the internal network card, and after the proxy server of the external network card receives the first request, the proxy server recognizes the NAT, and the media server of the external network card receives the streaming media. SBC2 monitors the external Network card, and when a packet containing a first request or a streaming media arrives at the external Network card, SBC2 can automatically identify NAT, which is Network-Address-Translation, i.e. Network Address Translation, and when all hosts using a local Address, i.e. a private IP Address, communicate with the outside, all hosts need to translate their local addresses into global IP addresses on the NAT router, so as to be able to connect to the internet. When the first request, the network data packet or the audio/video data packet flows into the external network card from the VoLTE relay 1 side through the WAN, the first request, the network data packet or the audio/video data packet is monitored by the SBC2, and when the network data packet and the audio/video data packet pass through the firewall, general NAT conversion can only modify three layers of IP, and cannot modify application layer IP, which may cause inconsistency between field addresses such as RTCP and the like in SDP in an SIP message and actual interworking addresses, and finally cause an address error or port failure of the SIP signaling and the interaction through the media stream, and cannot correctly establish signaling communication, so the application layer IP needs to be modified after the SBC2 monitors the SIP signaling. If the IP address in the first request is modified correctly, the subsequent PBX3 can obtain the correct IP address, so that the media stream can reach the seat application, i.e. the terminal, corresponding to the correct IP address.

SBC2 obtains the IP address that needs to be loaded. After receiving the first request, the load balancing module in the SBC2 calculates, by using the resource usage data of the current node of the SBC2, the maximum remaining available resource based on the resource usage data, and outputs the IP address corresponding to the maximum remaining available resource. And if the maximum remaining available resources are multiple, randomly outputting the IP address corresponding to one of the maximum remaining available resources. For example, when the SBC2 has a load balancing server built therein, and the first request is sent to the load balancing server of the SBC2, the load balancing server will obtain the resource usage of each media server on the current SBC2 node, and follow the rule that the maximum remaining available resource algorithm allocates preferentially and the same remaining available resources are allocated randomly, so as to calculate out that the IP address in the SDP needs to be adjusted to the correct IP address, so that media signaling can interact normally and reasonably. The SDP is called Session-Description-Protocol, which is a Protocol for describing a Session, and is mainly used for media negotiation between two Session entities. SIP is responsible for establishing and releasing sessions, which typically contain related media such as video and audio. The media data is described by SDP. SDP is not used alone, and when it is used in conjunction with SIP, SDP is put into the body of the SIP protocol. The SDP is used to perform media negotiation when a session is established, so that both parties can determine the media capabilities of the other party and exchange media data. After recognizing the NAT, the SBC2 replaces or replaces the IP address that cannot be successfully communicated with in the original first request with the calculated IP address, which is beneficial to finally forwarding the request to the PBX3, so that the PBX3 can obtain the correct IP address, and the media stream can reach the seat application.

And generating a type mark corresponding to the type content based on the preset type content analyzed from the first request, wherein the type content comprises video, audio or pictures. With reference to fig. 3, the SBC2 extracts the INVITE message in the first request, and the SBC2 determines whether the request is a voice request or a video request according to the content in the INVITE message, i.e., identifies the type content of the media stream corresponding to the first request, and matches the type tag from the preset tag target according to the type content. The type content can be voice or video, the type mark can be a mark corresponding to the voice or video, such as < voice > or < video >, the content of the streaming media is marked according to the content in the invitation information, the streaming media can be identified without reading the content of the streaming media, and the processing speed of the algorithm is improved.

Returning to fig. 2, the SBC2 makes the media service module receive the streaming media corresponding to the first request according to the first request received by the load balancing module, and then generates the SIP signaling that can be recognized by the PBX3 based on the empty SIP template according to the calculated correct IP address, the type tag, and the content of the first request. And generating a SIP communication request according to the SIP signaling and the type content, and sending the SIP communication request to the SIP contact center. When the load balancing module obtains the first request, the SBC2 further obtains a codec list supported by the calling party that issued the first request, matches and compares the SDP in the SIP signaling with parameters in the codec list, negotiates a coding mode matching the calling party according to a coding sequence provided in the SDP, and transcodes the streaming media based on a result that transcoding is required after negotiation. And performing transcoding operation in the process of generating the SIP signaling, wherein transcoding is not required if the calling party and the called party both support the same media format, and transcoding is required if the calling party and the called party do not support the same media format. When the step is specifically implemented, when the VoLTE relay 1 sends the first request to the external network card of the SBC2, the SIP message, that is, the first request carries the voice code supported by the SDP, and the SBC2 performs the coding negotiation inside. Firstly, the media server of the SBC2 matches and compares the SDP in the SIP message with the parameters provided by the codec list supported by the calling party, and negotiates to confirm the encoding mode of the calling party according to the encoding sequence provided in the SDP, at this time, the media server of the SBC2 sends the content of the first request to the called party, assuming that the encoding format of the calling party is PCMA, and if the codec list supported by the called party also supports the format of PCMA, the media server of the SBC2 does not need transcoding. However, if the encoder list of the called party only supports the format of the PCMU, the encoding information in the SDP message returned by the called party only carries the PCMU encoding, and the media server of the SBC2 needs to transcode as the intermediate media.

After the SIP communication request reaches the PBX3, the PBX3 performs a voice response interpretation on the SIP communication request, identifies a type tag during the voice response interpretation, and the PBX3 identifies streaming media from the SIP communication request according to the type tag and plays the streaming media according to a playing method corresponding to the type tag. Wherein, the step of playing comprises playing video or playing pictures and audio. The video type can be divided into a pure video type and a picture type and a voice type by using the type mark, and the pure video can be directly identified and played, so that the effect of video IVR is realized; the picture + speech type is a composite mode, i.e. a video effect is achieved by a combination of a separate picture and a separate audio. And before the step of playing the pictures and the voice, setting the preset duration of the pictures, the preset frequency of the audio and the preset track of the playing media in the type marks. When the picture and the voice are played, the picture is displayed according to the preset track until the preset duration and the audio is played for the preset frequency. The time length of picture display and the frequency of audio playing can be set in the type mark, when the type mark is identified, the picture with fixed time length and the frequency of circularly playing the audio can be displayed, and finally the effect of dynamically allocating the combination of the picture and the audio is achieved.

In the specific implementation of this step, an IVVR visual process server is built in the PBX3, and the IVVR visual process server provides an IVR interpreter based on the vxml3.0 standard, and newly adds an identifier for the < video > tag. Resource parameters and resource types such as png or mp4 can be set in the < video > tag, and a user can use the < video > tag to divide video types into two types, namely mp4 and picture + voice. Audio and video in mp4 format can be directly recognized and played on the media server of PBX3, thereby achieving the effect of video IVR. The picture + voice type is a synthesis mode, and a video effect is presented by combining a separate picture and a separate audio. The IVVR visualization process server can set the duration of picture display and the frequency of audio playing in the < video > tag, and when the media server of the PBX3 recognizes the < video > tag, the media server will display the picture with fixed duration and the frequency of playing audio cyclically in the seat application, so as to achieve the effect of dynamically allocating the combination of the picture and the audio. The IVVR visual process server can also set the timeout duration in the < video > tag, and can realize the effect of ending the session in a specific conversation scene or achieving no response after timeout. Meanwhile, the IVVR visual process server is consistent with the traditional IVR process server in the execution process, and also supports key interruption and voice interruption, and also supports functions such as key interaction and the like.

The interaction between the user terminal as the calling party and the seating application as the called party may further comprise: the SBC2 obtains an interactive node input by a caller, the caller can select a video voice from the interactive node to enter an SIP contact center, then the caller can input information into the PBX3 from the interactive node, the PBX3 parses a preset target instruction from the input information, the PBX3 accesses the caller according to the target instruction and waits for a first request or an invitation message from the first request, and the PBX3 receives the invitation message and then connects the caller to a called party corresponding to an IP address.

The implementation process of the embodiment of the application is as follows:

a calling party where a user is located enters an SIP contact center using a PBX3 through an SBC2 by dialing an incoming line number provided by a VoLTE relay 1, wherein the SBC2 acquires an IP address of a PBX3 to be loaded according to a first request sent by the calling party through the VoLTE relay 1, analyzes preset type content from the first request, and then marks the type content again, so that a subsequent SIP contact center can directly identify the type content in streaming media, and sends an SIP communication request generated according to newly generated SIP signaling and the streaming media to the SIP contact center, and the PBX3 of the SIP contact center can identify the streaming media sent by the VoLTE relay 1 through the SIP communication request, and an intranet which cannot support VoLTE voice and video communication can realize VoLTE voice and video communication. The IVVR visual process server in PBX3 will identify the process that the caller chooses to enter. The IVVR visual process server can analyze the node interaction selected by the user in the previous voice IVR guide process through the calling party, when the user selects man-transfer working hours in the IVR guide process, the IVVR visual process server can analyze manual statements transferred by VXML scripts, at the moment, the IVVR visual process server can enable a media server of the SBC2 to transfer the calling of the calling party to a routing point of a corresponding application server through a console command form, after an application platform of a seat application receives an INVITE message of the routing point, the application server can search the logged-in seat application in an idle state and a seat list with audio and video processing capacity in a system, the SBC2 matches a configured ACD distribution algorithm, namely a distribution algorithm of first idle service and maximum idle time to generate a correct IP address and generate an SIP communication request based on the SIP communication request, PBX3 routes the call from the secondary primary caller to the appropriate seat. When the seat application receives the INVITE request, the call of the main caller arrives at the seat application, at this time, the seat application returns a message to the media server of the SBC2, and at the same time, the seat application displays an audio and video incoming call, and at this time, when an ANSWER button on the seat application is clicked, the media server of the SBC2 initiates an ANSWER request, so that the seat application and the main caller finally establish a communication channel for audio and video communication. In the process of audio and video conversation, the seat application can use a stream transfer function, namely, a current ongoing video conversation is pushed to a client terminal of a main calling party, the client terminal of the main calling party can play the received stream video, the seat application keeps the audio and video conversation with the main calling party, and after the stream video is played, the conversation is recovered again. When the agent application uses the stream pushing function, the media server seat of the SBC2 receives a stream pushing request initiated by the agent application, where the request includes the information of the audio/video path of the stream pushing and the client terminal of the main caller, and the media server of the SBC2 maintains the call of the agent application, and makes the client terminal play the formulated audio/video, so as to achieve the stream pushing effect. And after the streaming video is played, bridging the seat application and the client terminal of the main calling party again, and recovering the audio and video conversation. The agent application can also use the screen sharing function to synchronously share the current screen desktop of the agent application to the client terminal of the main calling party. The agent application transmits the local desktop content to the media server of the SBC2 in a streaming video manner, and then the media server of the SBC2 shares the streaming video to the client terminal of the main caller. When the seat application clicks the screen sharing function, the seat application starts to acquire a local desktop by using a specific frequency and transmits acquired streaming data to the media server of the SBC2, and the media server of the SBC2 controls the client terminal of the main calling party to play the streaming video through the SIP protocol in real time, so as to achieve the screen sharing function.

The embodiment of the application also discloses a system and a system for the VoLTE relay to access the SIP contact center, wherein the system runs a computer program of any one method for the VoLTE relay to access the SIP contact center.

The embodiment of the application also discloses a computer storage medium which stores a computer program capable of being loaded by a processor and executing any one of the methods for accessing the VoLTE relay to the SIP contact center.

The above embodiments are preferred embodiments of the present application, and the protection scope of the present application is not limited by the above embodiments, so: all equivalent changes made according to the structure, shape and principle of the present application shall be covered by the protection scope of the present application.

Claims

1. A method for accessing a VoLTE relay to an SIP contact center is characterized in that: the method comprises the following steps:

acquiring an IP address after load distribution based on receiving a first request from a VoLTE relay (1);

and sending the SIP communication request to an SIP contact center.

2. The method for VoLTE relay access to SIP contact center according to claim 1, wherein: the step of receiving a first request from a VoLTE relay (1) further comprises:

monitoring an external network card;

and the media server of the external network card receives the streaming media.

3. The method for VoLTE relay access to SIP contact center according to claim 1, wherein: the step of obtaining the loaded IP address further includes the steps of:

4. The method for VoLTE relay access to SIP contact center according to claim 1, wherein: in the step of generating a type tag corresponding to the type content based on parsing a preset type content from the first request, the method further includes:

extracting an invitation message in the first request;

5. The method for VoLTE relay access to SIP contact center according to claim 1, wherein: the step before the first request from the VoLTE relay (1) is received further comprises:

6. The method for VoLTE relay access to SIP contact center according to claim 1, wherein: in the step of generating the SIP signaling according to the IP address, the type flag, and the first request, the method further includes:

7. The method for VoLTE relay access to SIP contact center according to claim 1, wherein: the method further comprises the following steps:

8. The method for VoLTE relay access to SIP contact center according to claim 7, wherein: the method further comprises the following steps:

9. A system for VoLTE relay access SIP contact center is characterized in that: a system running a computer program with a method for VoLTE relay access to a SIP contact center as claimed in any of claims 1 to 8.

10. A computer storage medium, characterized in that: a computer program loadable by a processor and adapted to perform the method of VoLTE relay access SIP contact center of any of claims 1-8.