CN-117041220-B - Voice networking service method based on SIP protocol

Abstract

The invention discloses a voice networking service method based on an SIP protocol, which belongs to the technical field of multimedia voice communication. The invention 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 an SIP protocol, the communication protocol between the DNS, the NMS and the VRS module adopts a TCP protocol, the communication protocol between the NMS module adopts a 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, and on the premise that a calling user calls a called user in a networking, the DNS module completes platform addressing and indicates the voice dispatching platform to establish call connection between the calling user and the called user through the NMS module. The invention can provide networking service among different voice dispatching platforms, can dynamically configure a routing path and has the functions of port convergence and the like. The invention provides an extended function interface, which greatly improves the flexibility and expansibility of networking.

Inventors

• ZHANG WEIZHENG
• ZHANG RUI
• HAN BO
• GUAN ZHENBO
• KONG ZHIFEI

Assignees

• 中国电子科技集团公司第五十四研究所

Dates

Publication Date

20260508

Application Date

20230918

Claims (4)

1. 1. A voice networking service method based on SIP protocol, comprising the steps of: step 1, constructing a voice networking system, wherein each networking unit in the voice networking system comprises a DNS module, an NMS module and a VRS module, the DNS module, the NMS module and the VRS module are connected through a TCP protocol, and the networking units form a tree topology structure by taking the respective DNS module as a route; When a voice dispatching platform is to be added into a networking, the voice dispatching platform sends a registration SIP signaling to an NMS module of a networking unit, and the NMS module adds the voice dispatching platform into the networking through authentication management; step 2, when a calling user calls a called user, the calling user sends a call request SIP signaling to a calling user voice dispatch platform; step 3, after receiving the call request SIP signaling, the calling party voice dispatching platform forwards the call request SIP signaling to an NMS module of the calling party networking unit; Step 4, after receiving the SIP signaling of the call request, the NMS module of the calling party networking unit judges whether the called party is a user needing to be routed, if not, the NMS module of the calling party networking unit directly executes step 5 and step 7-step 10, and the calling party networking unit and the called party networking unit are the same networking unit, if needing to be routed, the DNS module of the calling party networking unit initiates a routing application to the DNS module of the upper level networking unit, and the DNS module of the upper level networking unit initiates addressing through the information of the called party to provide an optimal routing path; step 5, the NMS module of the calling party networking unit inquires available media stream forwarding ports after convergence processing from the VRS module of the calling party networking unit, and re-encapsulates the SIP signaling message; step 6, the DNS module of the calling party networking unit guides the NMS module of the calling party networking unit to send the call request SIP signaling to the NMS module of the called party networking unit according to the route path planned by the DNS module of the superior networking unit; step 7, the NMS module of the called party networking unit sends the SIP signaling of the call request to the dispatching platform of the called Fang Huayin, and instructs the dispatching platform of the called Fang Huayin to establish the call with the called user; step 8, the called user returns the call confirmation signaling and ringing signaling information to the NMS module of the calling networking unit according to the route path through the NMS module of the called networking unit to complete the establishment of the whole call link; Step 9, the calling user and the called user send the media stream information to respective voice dispatching platforms, and the two-party voice dispatching platforms forward the media stream data to VRS modules of respective networking units; and 10, the VRS module of the two-party networking unit indicates the forwarding of the media stream data between the two-party voice scheduling platforms according to the planned routing path.
2. 2. The voice networking service method based on the SIP protocol according to claim 1, wherein the DNS modules communicate with each other through a TCP protocol to share routing information among the DNS modules, the DNS modules communicate with the NMS modules through the TCP protocol to indicate routing paths among the NMS modules, and the NMS modules establish communication connection according to the routing paths planned by the DNS modules through the TCP protocol to encapsulate the SIP signaling messages into TCP protocol messages to be transmitted with each other, so that the signaling messages conforming to the SIP protocol can be forwarded freely among the NMS modules.
3. 3. The voice networking service method based on the SIP protocol as set forth in claim 1, wherein the calling subscriber is an intra-group subscriber and the called subscriber is an extra-group subscriber, or the calling subscriber is an extra-group subscriber and the called subscriber is an intra-group subscriber, or both the calling subscriber and the called subscriber are intra-group subscribers.
4. 4. The voice networking service method based on the SIP protocol according to claim 1, wherein the DNS module is used for addressing and route optimization of the voice scheduling platform, providing addressing service for each node in the networking unit, and providing an optimal route path; The NMS module is used for registering the analysis of the SIP signaling and the authentication management of the registration information, completing the signaling scheduling and forwarding based on the SIP protocol according to the routing rule, guiding the VRS module to complete the media scheduling, and completing the cascade management and cascade scheduling among the networking units according to the requirements; The VRS module is used for completing the load and forwarding optimization of the media stream information of the voice scheduling platform, completing the multiplexing forwarding of the audio stream and the data stream and simultaneously providing port convergence service.

Description

Voice networking service method based on SIP protocol Technical Field The invention belongs to the technical field of multimedia voice communication, in particular to a voice networking service method based on an SIP protocol, which is applicable to an IP dispatching platform system and a voice networking management system based on the SIP protocol (Session Initiation Protocol: session initiation protocol). Background With the explosive growth of media information and the acceleration of the pace of people's work and life, a communication mode using digital text as a medium is difficult to adapt to a new work and life mode of people, and the demand for a more intelligent and reliable voice media communication mode is also increasing. Many special work scenarios cannot use existing audio network office software in the market in view of their application needs for external security. Different scenes can develop and use voice scheduling platforms meeting respective requirements to complete voice office, which prevents secret voice business between the scenes from going and going, so that reliable networking transmission service meeting different closed network environments is needed to realize interconnection and interworking between different platforms. The multimedia service network based on IP (Internet Protocol, network protocol) is widely used as a mature and reliable scheme in many scenes, and can integrate the existing urban fixed telephone and mobile telephone and add multimedia data service and other value-added services on the basis. The multimedia service network is widely focused by virtue of the characteristics of openness, simplicity, strong operability and the like, and rapidly expands and generates a plurality of extension services, thereby effectively reducing the operation and maintenance cost and the construction investment and providing conditions for media interconnection and interworking based on the IP protocol network. The SIP protocol is used as an IP voice control protocol from the Internet, has the characteristics of flexibility, reliability, convenient implementation and strong expansibility, becomes a main stream means of the current multimedia service network, and provides technical possibility for networking transmission between different closed environments. The SIP protocol is a text-based protocol similar to HTTP, which can be encapsulated by TCP or implemented by UDP. The method inherits the characteristics of simple, open and flexible internet protocol, ensures manageability for users and sessions, and can effectively lighten network core load. Meanwhile, compared with other protocols, the SIP also increases the signaling and QoS control requirements, so that the function implementation is richer and more reliable. The SIP protocol specifies that messages fall into two broad categories, one is a request message sent by a client to a server and the other is a response message returned by the server. In order to meet the demands of business confidentiality, the networking system needs to have a certain sealing performance, and can still work stably under the condition of not accessing the internet. Because of the difference of voice scheduling platforms among different working scenes, networking services need to be capable of supporting information interaction with different platforms, which presents challenges for compatibility of networking services. Firewall policies generally exist in application scenes with security, and policies of open communication ports are different among different scenes, so that networking needs to be capable of dynamically configuring port ranges and having a port convergence function. Meanwhile, the closed networking at present needs to meet the requirement of dynamic link configuration, can realize peer-to-peer networking and upper and lower-level networking, and can provide interfaces for increasing the requirements of personalized scenes, so that the whole networking process is more flexible, controllable and safe, and the challenges are presented to the traditional networking scheme. The traditional networking mode generally only supports one voice scheduling platform, and the working mode is as follows: the voice dispatching platforms are manually matched with the information of the other party; A calling user sends a call invitation signaling to a voice dispatching platform; the voice scheduling platform sends the voice scheduling information to the target voice scheduling platform through the matching relation; the target voice dispatch platform forwards the call invitation signaling to the called user. The communication mode between the calling user and the called user is realized based on a voice dispatching platform, networking is completed through the service of each platform, the problem of cross-platform media stream communication is solved to a certain extent, and the problem still exists that 1. The mode can not adapt to the cross-