A voice group service method based on SIP protocol

Abstract

The application discloses a voice networking service method based on SIP protocol and belongs to the technical field of multimedia voice communication. The application firstly constructs a DNS module, an NMS module and a VRS module, wherein the communication protocol between the NMS module and an external voice dispatching platform adopts the SIP protocol, the communication protocol between the DNS, the NMS and the VRS module adopts the TCP protocol, the protocol adopted by the communication between the NMS modules is the TCP protocol, the voice dispatching platform is used for accessing a user terminal, is connected with the user terminal through the SIP protocol and is connected with the NMS module through the SIP protocol; under the premise that a calling user calls a called user in a networking, the platform addressing is completed by the DNS module, and the voice dispatching platform is instructed by the NMS module to establish the call connection between the calling user and the called user. The application can provide the networking service between different voice dispatching platforms, can dynamically configure a routing path and has functions such as port convergence. The application provides an extended function interface, greatly improves the flexibility and expansibility of the networking.

Description

Technical Field

[0001] This invention belongs to the field of multimedia voice communication technology, and specifically refers to a voice networking service method based on the SIP protocol, which is applicable to IP scheduling platform systems and voice networking management systems based on the SIP protocol (Session Initiation Protocol). Background Technology

[0002] With the explosive growth of media information and the accelerating pace of people's work and life, communication methods based on digital text are no longer adequate for people's new work and lifestyles, and the demand for more intelligent and reliable voice media communication methods is deepening. Many special work scenarios, due to their confidentiality requirements, cannot use existing audio network office software on the market. Different scenarios develop and use voice dispatch platforms that meet their respective needs to complete voice-based office work, which hinders confidential voice communication between such scenarios. Therefore, a reliable networking transmission service is needed to achieve interconnection and interoperability between different platforms in different closed network environments.

[0003] IP (Internet Protocol)-based multimedia service networks, as a mature and reliable solution, have been widely used in many scenarios. They can integrate existing landline and mobile phones, and add multimedia data services and other value-added services. Due to their openness, simplicity, and strong maintainability, these multimedia service networks have attracted widespread attention and rapidly expanded to generate many extended services, effectively reducing operation and maintenance costs and construction investment, and providing conditions for media interconnection and interoperability based on IP protocol networks.

[0004] SIP, an IP voice control protocol originating from the Internet, is characterized by its flexibility, reliability, ease of implementation, and strong scalability. It has become the mainstream method for multimedia service networks, providing technical possibilities for network transmission between different closed environments. SIP is a text-based protocol similar to HTTP, which can be encapsulated using TCP or implemented using UDP. It inherits the simplicity, openness, and flexibility of Internet protocols, ensuring manageability of users and sessions while effectively reducing the network core load. Compared to other protocols, SIP also adds signaling and QoS control requirements, making its functionality richer and more reliable. The SIP protocol defines messages as divided into two main categories: request messages sent by clients to servers and response messages returned by servers.

[0005] To meet business confidentiality requirements, the networking system needs to possess a certain degree of isolation, enabling stable operation even without internet access. Due to the differences in voice dispatch platforms across various work scenarios, the networking service needs to support information exchange with different platforms, posing a challenge to its compatibility. Confidential applications typically employ firewall policies, and the policies for opening communication ports differ across scenarios; therefore, the network needs to dynamically configure port ranges and possess port convergence capabilities. Simultaneously, current closed-loop networks also need to support dynamically configurable links, enabling both peer-to-peer and hierarchical networking, and providing interfaces to add interfaces for customized scenarios, making the entire networking process more flexible, controllable, and secure. This presents a challenge to existing traditional networking solutions.

[0006] Traditional networking modes typically support only one voice dispatching platform, and their operation is as follows:

[0007] Manually connect each other's information between voice dispatch platforms;

[0008] The calling user sends a call invitation signal to the voice dispatch platform;

[0009] The voice dispatch platform sends the data to the target voice dispatch platform through the matching relationship;

[0010] The target voice dispatch platform forwards the call invitation signal to the called user.

[0011] This communication method between the calling and called users is based on a voice dispatching platform. It completes the network through the services of each platform, solving the problem of cross-platform media stream communication to some extent. However, it still has the following problems: 1. This method cannot adapt to cross-platform needs. Networking between different dispatching platforms requires targeted development and adaptation, increasing development costs. 2. This method cannot dynamically manage the connectivity between network links. When cross-scenario media stream transmission is not needed, it will cause resource waste, and keeping the link constantly open poses security risks. 3. This method only satisfies networking between dispatchers at the same level, lacking hierarchical networking relationships. Higher-level units cannot monitor the connection and call status of lower-level units. 4. It lacks a port convergence strategy, failing to solve security issues in cross-platform communication. 5. The platforms are mutually isolated, unable to meet the needs of personalized customization. Due to these problems, this solution is not suitable for current networking requirements. Summary of the Invention

[0012] To address the aforementioned issues, this invention provides a voice networking service method based on the SIP protocol, applicable to IP scheduling platforms and voice networking management systems based on the SIP protocol (Session Initiation Protocol). It supports bridging various voice scheduling platforms and can also be used as a replacement for voice scheduling platforms.

[0013] To achieve the above objectives, the technical solution adopted by the present invention is as follows:

[0014] A voice networking service method based on the SIP protocol includes the following steps:

[0015] Step 1: Construct a voice networking system. Each networking unit in the voice networking system includes a DNS module, an NMS module, and a VRS module. The DNS module, NMS module, and VRS module are connected via TCP protocol. The networking units form a tree topology by using their respective DNS modules as routes.

[0016] When a voice dispatch platform wants to join the network, the voice dispatch platform sends a registration SIP signaling to the NMS module of a network unit. The NMS module then adds the voice dispatch platform to the network through authentication and authorization management.

[0017] Step 2: When a calling user calls a called user, the calling user sends a call request SIP signaling to the calling party's voice dispatch platform;

[0018] Step 3: After receiving the call request SIP signaling, the calling party's voice dispatch platform forwards the call request SIP signaling to the calling party's network unit's NMS module;

[0019] Step 4: After receiving the call request SIP signaling, the NMS module of the calling party's network unit determines whether the called user is a user that needs to be routed. If no routing is required, it means that the calling user and the called user are under the same voice dispatch platform, and steps 5 and 7-10 are executed directly. At this time, the calling party's network unit and the called party's network unit are the same network unit. If routing is required, the DNS module of the calling party's network unit initiates a routing request to the DNS module of the superior network unit. The DNS module of the superior network unit initiates addressing through the called user information and provides the optimal routing path.

[0020] Step 5: The NMS module of the calling party's networking unit queries the VRS module of the calling party's networking unit for the available media stream forwarding port after convergence processing, and re-encapsulates the SIP signaling message.

[0021] Step 6: The DNS module of the calling party's network unit guides the NMS module of the calling party's network unit to send the call request SIP signaling to the NMS module of the called party's network unit according to the routing path planned by the DNS module of the superior network unit.

[0022] Step 7: The NMS module of the called party's network unit sends the call request SIP signaling to the called party's voice dispatching platform, instructing the called party's voice dispatching platform to establish a call with the called user;

[0023] Step 8: The called party user returns the call confirmation signaling and ringing signaling messages to the calling party's network unit's NMS module through the called party's network unit's NMS module according to the routing path, thus completing the establishment of the entire call link;

[0024] Step 9: The calling and called users send media stream information to their respective voice dispatch platforms, and the two voice dispatch platforms forward the media stream data to the VRS module of their respective network units;

[0025] Step 10: The VRS modules of the two networking units forward media stream data between the two voice dispatching platforms according to the planned routing path.

[0026] Furthermore, the DNS modules communicate with each other via TCP protocol to share routing information; the DNS module and the NMS module communicate via TCP protocol, with the DNS module indicating the routing path between the NMS modules; the NMS modules establish communication connections via TCP protocol according to the routing path planned by the DNS module, encapsulate SIP signaling messages into TCP protocol messages for mutual transmission, thereby freely forwarding SIP-compliant signaling messages between the NMS modules.

[0027] Furthermore, the calling user is a user within the group, and the called user is a user outside the group; or the calling user is a user outside the group, and the called user is a user within the group; or both the calling user and the called user are users within the group.

[0028] Furthermore, the DNS module is used for addressing and routing optimization of the voice dispatching platform, providing addressing services for each node within the networking unit and providing the optimal routing path;

[0029] The NMS module is used for parsing and authenticating registration information of SIP signaling, and performs signaling scheduling and forwarding based on the SIP protocol according to routing rules. It guides the VRS module to perform media scheduling and performs cascading management and scheduling between network units as needed.

[0030] The VRS module is used to optimize the load and forwarding of media stream messages on the voice scheduling platform, perform multiplexing and forwarding of audio streams and data streams, and provide port convergence services.

[0031] The beneficial effects of this invention are as follows:

[0032] 1. This invention can provide networking services between different voice dispatching platforms, can dynamically configure routing paths and has functions such as port convergence.

[0033] 2. This invention provides an extended function interface, which greatly improves the flexibility and scalability of networking.

[0034] 3. This invention is applicable to IP scheduling platforms and voice networking management systems based on the SIP protocol. It supports bridging various voice scheduling platforms and can also be used as a replacement for voice scheduling platforms. Attached Figure Description

[0035] Figure 1 This is a schematic diagram of the structure of a networking unit in the voice networking service method of the present invention.

[0036] Figure 2 This is a schematic diagram of the cascaded structure of the networking units in the voice networking service method of the present invention.

[0037] Figure 3 This is a flowchart of establishing a cross-platform session in an embodiment of the present invention.

[0038] Figure 4 This is a schematic diagram of the port convergence process in an embodiment of the present invention.

[0039] Figure 5 This is a flowchart of network routing path planning in an embodiment of the present invention. Detailed Implementation

[0040] The preferred embodiments of the present invention will be described below with reference to the accompanying drawings. The preferred embodiments described herein are for illustration and explanation only and are not intended to limit the present invention.

[0041] A voice networking service method based on the SIP protocol is disclosed, which implements a DNS (Domain Name System) module, an NMS (Network Management Service) module, and a VRS (Voice Route Service) module. The DNS module is used for platform addressing, routing optimization strategies, and providing addressing services for various basic modules, service modules, and terminal nodes within the voice networking service, providing optimal routing paths. The VRS module optimizes the load balancing and forwarding of media stream messages on the platform, and can also perform multiplexing and forwarding of audio and data streams, while providing port convergence services. The NMS module is used for parsing and authenticating registration information of SIP signaling, and performs SIP-based signaling scheduling and forwarding according to routing rules, guiding the VRS module to perform media scheduling, and performing cascading management and scheduling between voice networking services as needed. The VRS module, NMS module, and DNS module communicate with each other via the TCP (Transmission Control Protocol).

[0042] The method includes the following steps:

[0043] Step 1: The voice dispatching platform sends a registration SIP signaling message to the NMS module, and the NMS module adds the voice dispatching platform to the network through authentication and authorization management.

[0044] Step 2: After the calling user calls the called user, the calling user sends a call request SIP signaling to the voice dispatch platform.

[0045] Step 3: After receiving the call request SIP signaling, the voice dispatching platform forwards the call request SIP signaling to the NMS module.

[0046] Step 4: After receiving the call request SIP signaling, the NMS module determines whether the called user is a user that needs to be routed.

[0047] Step 5: The call request needs to be routed. The NMS module initiates a route request to the DNS module, and the DNS module initiates address lookup in the called user information to provide the optimal route path.

[0048] Step 6: The NMS module queries the VRS module for available media stream forwarding ports and re-encapsulates the SIP signaling message.

[0049] Step 7: The DNS module instructs the NMS module to send the encapsulated SIP signaling to the target NMS module.

[0050] Step 8: The target NMS module sends the call request SIP signaling to the target voice dispatching platform, instructing the target voice dispatching platform to establish a call with the called user.

[0051] Step 9: The called user returns a call confirmation message and a ringing message to the NMS module according to the routing path. The NMS module then instructs the voice dispatching platform to establish a call with the terminal.

[0052] Step 10: The calling user sends the media stream information to the voice dispatch platform, which then forwards the media stream data to the VRS module.

[0053] Step 11: The VRS module forwards media stream data between voice scheduling platforms according to the planned routing path.

[0054] In the above method, the voice networking service and the voice dispatching platform are independent IP traffic systems. The voice networking service and the voice dispatching platform communicate using the SIP protocol. The voice networking service and the voice dispatching platform complete directory lookup, network link authorization, network link disconnection, and personalized customization services through extended MESSAGE messages.

[0055] During cross-platform call setup, this method offers two cascading options: hard connection and soft connection. The hard connection method requires the voice networking service to be configured in peer-to-peer interconnection mode. The two peer-to-peer voice networking services establish communication via TCP, and this connection does not age over time. The soft connection method requires the voice management service to be configured in multi-level interconnection mode. A higher-level voice networking service is added to the two peer-to-peer voice networking services that need to establish communication. The higher-level networking service dynamically manages the communication status of the two peer-to-peer networking service links.

[0056] In the cascading scheme, during call establishment, the calling user sends an invitation message (INVITE message) to the voice dispatch platform. The voice dispatch platform then forwards the invitation message (INVITE message) to the voice networking service. The voice networking service, through its NMS module, sends a redirected and encapsulated TCP message to the destination networking service's NMS module. The destination networking service's NMS module re-encapsulates the parsed key information into an invitation message (INVITE message) and sends it to the voice dispatch platform. Upon receiving the invitation message (INVITE message), the called user returns an acknowledgment message (200 OK message) and a ringing message (180 RINGING message). The calling user, upon receiving the acknowledgment message (200 OK message), returns an ACK acknowledgment message, completing the entire call connection process. The DNS module manages the routing paths between NMS modules, and the VRS is responsible for media stream communication and port convergence services. In hard-connected mode, there is a strong pointing relationship between peer voice networking services; once configured, a TCP-based binding relationship is established immediately. In soft connection mode, the DNS module of the peer voice networking service queries the routing information from the upper-level DNS module, which can realize dynamic configuration of routing paths. In addition, an aging policy is added, which will automatically disconnect the routing relationship between networking services if the link is not used for a long time.

[0057] Here is a more specific example:

[0058] A voice networking service method based on the SIP protocol, wherein the networking unit is as follows: Figure 1 As shown, the core functions of the voice networking service are implemented through three modules: NMS, DNS, and VRS. These three modules communicate with each other via the TCP protocol.

[0059] Voice networking service cascading structure such as Figure 2As shown, this soft-connection cascading method enables the management of routing status between Voice Networking Service 1 and Voice Networking Service 2 through voice networking services, achieving a more flexible configuration of routing status. When User 1 needs to initiate a call to User 2, User 1 sends an INVITE signaling message to NMS module 1 of Voice Networking Service 1 through the voice dispatching platform. NMS module 1 parses the INVITE signaling message and sends the information to DNS module 1 via TCP protocol to query the routing table. If the destination networking service is not in the routing table of DNS module 1, then DNS module 1 queries the routing information from the voice networking service DNS module via TCP protocol. Based on the routing information provided by the DNS module, a SIP routing path is established between Voice Networking Service 1 and Voice Networking Service 2. The NMS module queries the VRS module for available media stream communication port numbers and re-encapsulates the INVITE signaling message. NMS module 1 sends the INVITE signaling message to NMS module 2 according to the planned routing path. NMS module 2 then sends the parsed INVITE signaling message to the called user 2 through the voice dispatching platform. It should be noted that during the forwarding process between NMS modules, key information values ​​such as FROM and TO in the SIP protocol will change dynamically as the signaling flows. The VRS module manages the SDP information in the SIP protocol.

[0060] Figure 3 This is a flowchart of cross-platform session establishment in an embodiment of the voice networking service method. The process includes the following steps:

[0061] Step 100: User 1 dials User 2's terminal number and sends the INVITE call signaling message to the voice dispatch platform 1.

[0062] Step 101: The voice dispatching platform 1 sends the INVITE call signaling to the NMS module 1 of the voice networking service 1 without discrimination.

[0063] Step 102: NMS module 1 parses the INVITE call signaling and sends the parsed data to DNS module 1 using the TCP protocol.

[0064] Step 103: DNS module 1 finds the path that needs to be routed and returns the route path to NMS module 1.

[0065] Step 104: NMS module 1 establishes a TCP communication connection with NMS module 2 using the queried routing path.

[0066] Step 105: NMS module 1 forwards the received INVITE call signaling to NMS module 2 according to the routing path.

[0067] Step 106: After re-encapsulating the FROM and TO fields of the INVITE signaling, NMS module 2 forwards the modified INVITE signaling to voice dispatching platform 2.

[0068] Step 107: The voice dispatch platform 2 forwards the received INVITE call signaling to user 2.

[0069] Step 108: After receiving the INVITE call invitation signaling, User 2 returns a 200 OK confirmation signaling to the voice dispatching platform 2.

[0070] Step 109: After receiving the 200 OK confirmation signal, the voice dispatch platform 2 modifies the 200 OK confirmation signal and forwards it to the NMS module 2.

[0071] Step 110: After receiving the 200 OK confirmation signal, NMS module 2 encapsulates the signal into a TCP message and forwards it to NMS module 1.

[0072] Step 111: After receiving the TCP message, NMS module 1 re-encapsulates it into SIP signaling and forwards it to voice dispatching platform 1.

[0073] Step 112: The voice dispatch platform 1 returns a 200 OK confirmation message to user 1.

[0074] Step 113: User 1 receives a 200 OK confirmation signal and returns an ACK confirmation signal to the voice dispatch platform 1.

[0075] Step 114: The voice dispatching platform 1 sends the ACK confirmation information to the NMS module 1.

[0076] Step 115: After receiving the ACK confirmation information, NMS module 1 re-encapsulates it and sends it to NMS module 2 according to the routing path.

[0077] Step 116: NMS module 2 sends the received ACK confirmation information to voice dispatch platform 2.

[0078] Step 117: The voice dispatch platform 2 sends an ACK confirmation message to user 2 to complete the establishment of the call link.

[0079] In the above voice networking method, the NMS module and the voice dispatching platform communicate via MESSAGE messages to complete status list queries or other personalized customization services.

[0080] Figure 4 This is a flowchart of the port convergence process during session establishment, used to adapt firewall policies for confidential application scenarios and resolve the issue of different open communication port policies across different scenarios. It includes the following steps:

[0081] Step 200: After receiving the INVITE signaling message, the voice dispatching platform 1 forwards the message to the NMS module 1 of the voice management service 1.

[0082] Step 201: NMS module 1 queries VRS module 1 for the media stream communication port.

[0083] Step 202: VRS module 1 returns the media stream communication port to NMS module 1, instructing NMS module 1 to modify the SDP handshake message of the INVITE signaling.

[0084] Step 203: NMS module 1 sends the re-encapsulated INVITE signaling message to NMS module 2.

[0085] Step 204: NMS module 2 forwards the INVITE signaling message to voice dispatching platform 2.

[0086] Step 205: After receiving the INVITE signaling message, the voice dispatching platform 2 returns a 200 OK confirmation message to the NMS module 2.

[0087] Step 206: NMS module 2 queries VRS module 2 for the media stream communication port.

[0088] Step 207: VRS module 2 returns the media stream communication port to NMS module 2, instructing NMS module 2 to modify the SDP handshake message of the INVITE signaling.

[0089] Step 208: NMS module 2 sends a re-encapsulated 200 OK signaling message to NMS module 1.

[0090] Step 209: NMS module 1 forwards the 200 OK signaling message to voice dispatching platform 1.

[0091] Step 210: After receiving the RTP media stream message, the voice dispatch platform 1 sends the message to the VRS module 1 according to the planned path.

[0092] Step 211: VRS module 1 sends the media stream message to VRS module 2 through the planned convergence port number.

[0093] Step 212: After receiving the media stream message, VRS module 2 forwards it to voice dispatch platform 2 through the planned path.

[0094] Furthermore, the VRS module port number convergence strategy restricts the allocated port numbers to a range, the upper and lower limits of which can be dynamically configured. Media stream communication between VRS modules is in full-duplex mode.

[0095] Figure 5 This is a flowchart of the routing plan during session establishment, including the following steps:

[0096] Step 300: The voice dispatching platform 1 forwards the received INVITE signaling message to the NMS module 1 of the voice networking service 1.

[0097] Step 301: NMS module 1 parses the INVITE signaling message and queries the DNS module 1 for routing information based on the parsed information.

[0098] Step 302: DNS module 1 queries the local routing table for routing information. If there is no routing configuration status locally, DNS module 1 queries the upstream DNS module for routing information.

[0099] Step 303: The upper-level DNS module returns the routing relationship to the lower-level DNS module 1.

[0100] Step 304: DNS module 1 sends the received routing information to NMS module 1.

[0101] Step 305: NMS module 1 forwards the INVITE signaling message to NMS module 2 according to the routing relationship planned by DNS module 1.

[0102] Step 306: NMS module 2 forwards the received INVITE signaling message to voice dispatch platform 2.

[0103] Furthermore, the communication protocol between the NMS module, DNS module, and VRS module is TCP, as are the communication protocols between the NMS modules and the DNS modules. Routing relationships for users within the local cluster are managed by the local DNS module, while cross-platform routing relationships are managed by the parent DNS module. This service can achieve a multi-level pyramid network structure through hierarchical configuration, thus meeting the needs of different scenarios.

Claims

1. A voice networking service method based on the SIP protocol, characterized in that, Includes the following steps: Step 1: Construct a voice networking system. Each networking unit in the voice networking system includes a DNS module, an NMS module, and a VRS module. The DNS module, NMS module, and VRS module are connected via TCP protocol. The networking units form a tree topology by using their respective DNS modules as routes. When a voice dispatch platform wants to join the network, the voice dispatch platform sends a registration SIP signaling to the NMS module of a network unit. The NMS module then adds the voice dispatch platform to the network through authentication and authorization management. Step 2: When a calling user calls a called user, the calling user sends a call request SIP signaling to the calling party's voice dispatch platform; Step 3: After receiving the call request SIP signaling, the calling party's voice dispatch platform forwards the call request SIP signaling to the calling party's network unit's NMS module; Step 4: After receiving the call request SIP signaling, the NMS module of the calling party's network unit determines whether the called user is a user that needs to be routed. If no routing is required, it means that the calling user and the called user are under the same voice dispatch platform, and steps 5 and 7-10 are executed directly. At this time, the calling party's network unit and the called party's network unit are the same network unit. If routing is required, the DNS module of the calling party's network unit initiates a routing request to the DNS module of the superior network unit. The DNS module of the superior network unit initiates addressing through the called user information and provides the optimal routing path. Step 5: The NMS module of the calling party's networking unit queries the VRS module of the calling party's networking unit for the available media stream forwarding port after convergence processing, and re-encapsulates the SIP signaling message. Step 6: The DNS module of the calling party's network unit guides the NMS module of the calling party's network unit to send the call request SIP signaling to the NMS module of the called party's network unit according to the routing path planned by the DNS module of the superior network unit. Step 7: The NMS module of the called party's network unit sends the call request SIP signaling to the called party's voice dispatching platform, instructing the called party's voice dispatching platform to establish a call with the called user; Step 8: The called party user returns the call confirmation signaling and ringing signaling messages to the calling party's network unit's NMS module through the called party's network unit's NMS module according to the routing path, thus completing the establishment of the entire call link; Step 9: The calling and called users send media stream information to their respective voice dispatch platforms, and the two voice dispatch platforms forward the media stream data to the VRS module of their respective network units; Step 10: The VRS modules of the two networking units forward media stream data between the two voice dispatching platforms according to the planned routing path.

2. The voice networking service method based on SIP protocol according to claim 1, characterized in that, DNS modules communicate with each other via the TCP protocol to share routing information. The DNS module and the NMS module communicate via the TCP protocol. The DNS module indicates the routing path between the NMS modules. The NMS modules establish a communication connection with each other via the TCP protocol according to the routing path planned by the DNS module, and encapsulate SIP signaling messages into TCP protocol messages for mutual transmission, thereby freely forwarding SIP-compliant signaling messages between the NMS modules.

3. The voice networking service method based on SIP protocol according to claim 1, characterized in that, The calling user is a user within the group, and the called user is a user outside the group; or the calling user is a user outside the group, and the called user is a user within the group; or both the calling user and the called user are users within the group.

4. The voice networking service method based on SIP protocol according to claim 1, characterized in that, The DNS module is used for addressing and routing optimization of the voice dispatching platform, providing addressing services for each node in the network unit and providing the optimal routing path; The NMS module is used for parsing and authenticating registration information of SIP signaling, and performs signaling scheduling and forwarding based on the SIP protocol according to routing rules. It guides the VRS module to perform media scheduling and performs cascading management and scheduling between network units as needed. The VRS module is used to optimize the load and forwarding of media stream messages on the voice scheduling platform, perform multiplexing and forwarding of audio streams and data streams, and provide port convergence services.

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