Anti-interference SIP telephone communication structure

Abstract

The utility model belongs to the technical field of telephone communication, in particular to an anti-interference SIP telephone communication structure, which comprises a core layer, an edge layer and a user layer, and is characterized in that the core layer comprises an SIP server and a media server and is used for processing SIP signaling and media streams; the edge layer comprises a firewall / NAT (Network Address Translation) traversal device and a load balancer, and is used for protecting system security and realizing load balancing; the user layer comprises an SIP telephone terminal and client software and is used for initiating and receiving telephone calls and managing information such as user accounts. According to the utility model, through multiple measures such as encryption and authentication mechanisms, call filtering and blocking functions, network quality detection and optimization, load balancing and fault-tolerant mechanisms and the like, the security and call quality of the system are effectively improved, unnecessary interference is reduced through call filtering and blocking, and the service life of the system is prolonged. And the availability and stability of the system are improved by using load balancing and fault tolerance, meanwhile, voice quality monitoring and coding transmission are optimized, and high-quality call experience is provided for users.

Description

Technical Field

[0001] The utility model relates to the technical field of telephone communication, in particular to an interference-proof SIP telephone communication structure. Background Art

[0002] With the development of communication technology, SIP telephone communication system has been widely used. However, in actual application, due to network interference, malicious attacks and other reasons, call quality often decreases or calls are interrupted. Utility Model Content

[0003] 1. Technical issues to be resolved

[0004] In view of the deficiencies in the prior art, the utility model provides an interference-proof SIP telephone communication structure, which solves the problem of call quality degradation or call interruption caused by network interference, malicious attacks and the like.

[0005] (II) Technical solution

[0006] In order to achieve the above-mentioned purpose, the utility model specifically adopts the following technical solutions:

[0007] An interference-proof SIP telephone communication structure adopts a layered architecture, including a core layer, an edge layer, and a user layer;

[0008] The core layer includes SIP servers and media servers, which are used to process SIP signaling and media streams;

[0009] The edge layer includes firewall / NAT penetration devices and load balancers to protect system security and achieve load balancing;

[0010] The user layer includes SIP phone terminals and client software, which are used to initiate and receive phone calls and manage user account information.

[0011] Furthermore, a TLS / SSL encrypted channel is established between the SIP server and the SIP phone terminal to encrypt the transmission of SIP signaling to prevent the signaling from being tampered with or stolen;

[0012] At the same time, SRP or other authentication mechanisms are used to authenticate users to ensure that only authorized users can access the system.

[0013] Furthermore, blacklist and whitelist functions are set up on the SIP server to allow or block calls from specific numbers;

[0014] In addition, call filtering rules can be set based on the caller's number, time, frequency and other factors to automatically filter out unnecessary calls and reduce interference.

[0015] Furthermore, the network quality detection function is implemented on the media server to monitor network delay, packet loss rate and other parameters during the call in real time;

[0016] When poor network quality is detected, the encoding method is automatically adjusted or parameters such as bit rate are reduced to optimize call quality.

[0017] Furthermore, a load balancer is deployed at the edge layer to distribute SIP requests to multiple SIP servers for processing, thus achieving load balancing and failover;

[0018] When a SIP server fails, the load balancer can automatically remove it from the system and forward requests to other normally operating SIP servers to ensure system availability and stability.

[0019] (III) Beneficial effects

[0020] Compared with the prior art, the utility model provides an interference-proof SIP telephone communication structure, which has the following beneficial effects:

[0021] The utility model effectively improves the security and call quality of the system through multiple measures such as encryption and authentication mechanism, call filtering and blocking function, network quality detection and optimization, load balancing and fault tolerance mechanism, reduces unnecessary interference through call filtering and blocking, and improves the availability and stability of the system by using load balancing and fault tolerance. At the same time, it optimizes voice quality monitoring and coding transmission to provide users with a high-quality call experience. BRIEF DESCRIPTION OF THE DRAWINGS

[0022] Figure 1 It is a structural schematic diagram of the utility model. DETAILED DESCRIPTION

[0023] The following will be combined with the drawings in the embodiments of the utility model to clearly and completely describe the technical solutions in the embodiments of the utility model. Obviously, the described embodiments are only part of the embodiments of the utility model, not all of the embodiments. Based on the embodiments in the utility model, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of the utility model.

[0024] Example

[0025] like Figure 1 As shown, an embodiment of the utility model proposes: an anti-interference SIP telephone communication structure, adopting a layered architecture, including a core layer, an edge layer and a user layer;

[0026] The core layer includes SIP servers, which are used to process SIP signaling, such as registration, call establishment, call transfer, etc.

[0027] Media server, which processes the actual voice or video stream and provides functions such as encoding and decoding and streaming;

[0028] The edge layer includes firewall / NAT traversal devices to protect the internal network from external attacks while ensuring that SIP signaling and media streams can correctly traverse the network address translation (NAT) device;

[0029] Load balancer, used to protect system security and achieve load balancing, distribute incoming SIP requests, and ensure high availability and load balancing of the system;

[0030] The user layer includes SIP phone terminals, which support the SIP protocol and can initiate and receive phone calls;

[0031] Client software used to make and receive phone calls and manage user account information, contact lists, call history, etc.

[0032] Among them, the SIP telephone communication structure can realize multiple modes of communication such as wireless, telephone line, network cable, etc. through the local area network as a carrier, thereby improving the scope of application and convenience of use.

[0033] Among them, a TLS / SSL encrypted channel is established between the SIP server and the SIP phone terminal to encrypt the transmission of SIP signaling to prevent the signaling from being tampered with or stolen;

[0034] At the same time, SRP or other authentication mechanisms are used to authenticate users to ensure that only authorized users can access the system.

[0035] Among them, set up blacklist and whitelist functions on the SIP server to allow or block calls from specific numbers;

[0036] In addition, call filtering rules can be set based on the caller's number, time, frequency and other factors to automatically filter out unnecessary calls and reduce interference.

[0037] Among them, the network quality detection function is implemented on the media server to monitor parameters such as network delay and packet loss rate during the call in real time;

[0038] When poor network quality is detected, the encoding method is automatically adjusted or parameters such as bit rate are reduced to optimize call quality.

[0039] Among them, a load balancer is deployed at the edge layer to distribute SIP requests to multiple SIP servers for processing, achieving load balancing and failover;

[0040] When a SIP server fails, the load balancer can automatically remove it from the system and forward requests to other normally operating SIP servers to ensure system availability and stability and implement fault-tolerant mechanisms such as backup servers and redundant equipment to cope with system failures or network interruptions.

[0041] Finally, it should be noted that the above description is only a preferred embodiment of the utility model and is not intended to limit the utility model. Although the utility model is described in detail with reference to the above embodiments, those skilled in the art can still modify the technical solutions recorded in the above embodiments or replace some of the technical features therein by equivalents. Any modification, equivalent replacement, improvement, etc. made within the spirit and principle of the utility model shall be included in the protection scope of the utility model.

Claims

1. An interference-proof SIP telephone communication structure, comprising a core layer, an edge layer and a user layer, characterized in that: The core layer includes a SIP server and a media server for processing SIP signaling and media streams; The edge layer includes firewall / NAT penetration equipment and load balancer, which are used to protect system security and achieve load balancing; The user layer includes SIP phone terminals and client software, which are used to initiate and receive phone calls and manage user account information.

2. The anti-interference SIP telephone communication structure according to claim 1, characterized in that: A TLS / SSL encrypted channel is established between the SIP server and the SIP telephone terminal to encrypt and transmit the SIP signaling to prevent the signaling from being tampered with or stolen.

3. The anti-interference SIP telephone communication structure according to claim 1, characterized in that: The SIP server is provided with blacklist and whitelist functions to allow or block calls from specific numbers.

4. The anti-interference SIP telephone communication structure according to claim 1, characterized in that: The media server is provided with a network quality detection module for real-time monitoring of parameters such as network delay and packet loss rate during a call.

5. The anti-interference SIP telephone communication structure according to claim 1, characterized in that: The load balancer is used to distribute SIP requests to multiple SIP servers for processing, thereby achieving load balancing and failover.

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