Railway electronic speed controller straight-through telephone protocol converter based on analog-to-digital conversion

Abstract

The invention relates to the field of communication, in particular to a railway electronic speed controller straight-through telephone protocol converter based on analog-to-digital conversion, which comprises seven parts, namely a network address conversion module, an Ethernet switching module, an SIP (session initiation protocol) processing module, a voice coding and decoding module, a TDM (time division multiplexing) switching module, an analog signaling processing module and a Z interface module, an IP telephone, an IAD or an ONU are connected through an IP network, the Z interface module provides a telephone line interface externally and is used for being connected with a digital dispatching switch, all the internal modules are matched with one another, conversion between analog signals and digital signals is completed, and the function of transmitting and dispatching direct-through telephone services through the IP network is achieved. The method has the advantages that the railway dispatching system is matched with the data network and the PON network, so that stable and reliable electronic speed controller straight-through telephone service is realized, and cost reduction and benefit increase are really realized; business protection is achieved through various protection means of a railway dispatching system, and business safety is guaranteed.

Description

Technical Field

[0001] The present application relates to the field of communications, in particular to the field of railway dispatching communications and passive optical networks. It is a protocol converter based on analog-to-digital conversion to solve the problem of railway electric dispatching direct telephone being carried on XGS-PON / GPON / network. Background Art

[0002] The railway dispatching communication system is a dedicated telephone communication system used for business communications between the dispatching command center, dispatchers at dispatching offices, and transportation production personnel within their jurisdiction. As a key component of railway communications, the railway digital dispatching system plays a vital role in ensuring efficient and safe railway transportation. The railway dispatching communication system consists of a trunk line dispatching communication system and a section dispatching communication system. Key equipment includes dispatching office switches, station dispatching switches, dispatching desks, duty desks, voice recorders, and network management equipment. Ancillary equipment includes optical distribution frames (ODFs), digital distribution frames (DDFs), and audio distribution frames. The system facilitates transportation command-related communications, including trunk line dispatching communications, section dispatching communications, station yard communications, inter-station communications, and inter-section communications. The in-station direct telephone line uses a standard dual-tone telephone connected to the common power port of the station dispatching switch via a local cable or narrowband access network system's ONU Z port. Users at the station or in the section can use this system to directly call the station or dispatching center operator, enabling instant calls without dialing. However, with the rapid development of communication technology, regional cables are aging and frequently disconnected, and PCM-based narrowband access network systems (ONUs) have ceased production, replaced by access systems such as GPON and XGS-PON. This transition has brought many challenges.

[0003] The existing railway digital dispatching system's direct telephone service is incompatible with GPON and XGS-PON systems, making it impossible to deliver direct telephone service to kiosks and users along the railway lines. This severely restricts the communication system requirements of railway modernization. First, traditional regional cable and narrowband access network systems have significant coverage limitations and cannot meet the high-bandwidth and high-speed office network requirements of users in areas such as kiosks and optical call poles. This leads to insufficient communication security in these critical areas, impacting railway operational efficiency and safety. Second, traditional regional cable and narrowband access networks fail to fully leverage the advantages of existing digital dispatching switches and railway GPON / XGS-PON systems, resulting in wasted equipment resources and increased operating costs, which runs counter to the modern enterprise management goal of reducing costs and increasing efficiency. Furthermore, traditional regional cable and narrowband access networks also have network security vulnerabilities, failing to effectively block communication ports other than voice protocols. This exposes the network to security risks and makes it vulnerable to unauthorized intrusions and attacks, threatening railway communication security. Summary of the Invention

[0004] The present application provides a railway electric dispatch direct telephone protocol converter and dispatching communication system based on analog-to-digital conversion, which is used to solve the problem that the electric dispatch direct telephone of the existing railway digital dispatching system cannot be transmitted to the booth and section users through GPON, XGS-PON and other systems.

[0005] The embodiment of the present application provides a railway electric dispatch direct telephone protocol converter based on analog-to-digital conversion, which is characterized by including: a network address translation module, an Ethernet switching module, a SIP protocol processing module, a voice codec module, a TDM switching module, an analog signaling processing module, a Z interface module and other components. The FE / GE (Fast Ethernet / Gigabit Ethernet) interface provided by the network address translation module is used to connect to a network telephone (IP phone), an integrated access device (IAD) or an optical network unit device (ONU). The network address translation module plays the role of internal and external address conversion. The Ethernet switching module serves as an internal Ethernet switch to connect the SIP protocol processing module, the voice codec module and the voice codec module. and a network address translation module. The SIP protocol processing module processes the voice communication protocol based on the IP network between the device and the IP phone, IAD or ONU. The telephone line interface provided by the Z interface module is used to connect to the digital scheduling switch. The TDM switching module serves as a time division switching network, connecting the Z interface module, the voice encoding and decoding module and the analog signaling processing module. The analog signaling processing module detects and processes the dual-tone multi-frequency signaling between the device and the scheduling switch in real time, and interacts with the SIP protocol processing module to complete the mutual conversion between the analog signaling and the SIP protocol. The voice encoding and decoding module realizes analog-to-digital or digital-to-analog conversion of the voice signal, and supports compression algorithms such as G.711, G.729 and AMR-WB.

[0006] As an improvement, the analog-to-digital conversion-based electric dispatch direct telephone protocol converter is arranged in the railway dispatching communication system.

[0007] As an improvement, the electric dispatch direct telephone protocol converter is connected to the railway digital dispatching switch through a telephone line, connected to the data network or PON network (passive optical network) through a digital Ethernet interface, and finally connected to the IP phone, IAD (integrated access device) or ONU (optical network unit) through the data network or PON network. The IP phone can be used as a direct telephone; the IAD or ONU can connect to the telephone as a direct telephone.

[0008] As an improvement, the PON network is mainly composed of an OLT (Optical Line Terminal), an optical splitter and an ONU (Optical Network Unit), and is mainly based on one of the three standard protocols of GPON, XGPON or XGS-PON.

[0009] As an improvement, the telephone protocol converter and the IP phone, IAD or ONU are connected using a 100M / 1000M / 10G network.

[0010] As an improvement, the voice codec module has a built-in programmable DSP chip that supports dynamic switching among three coding formats: G.711, G.729 and AMR-WB.

[0011] As an improvement, the analog signaling processing module can detect or generate DTMF (dual tone multi-frequency) analog signaling signals, which can realize the function of talking to the dispatching desk / duty desk without dialing when the phone is off-hook.

[0012] As an improvement, the described analog-to-digital conversion-based telephone protocol converter is characterized in that the SIP (Session Initiation Protocol) protocol is used between the IP phone, IAD or ONU.

[0013] As an improvement, dispatching the direct call includes the following steps:

[0014] S1: The dispatch office initiates a call request through the dispatch desk / duty desk, which is then sent to the direct telephone protocol converter via the station dispatch switch.

[0015] S2: The direct telephone protocol converter receives analog voice signals through the Z interface and distributes them to the analog signaling processing module through the TDM (time division multiplexing) switching module. The analog signaling processing module analyzes the voice signals in real time and parses the dialing information.

[0016] S3: The analog signaling processing module sends the parsed dialing information to the SIP protocol processing module;

[0017] S4: The SIP protocol processing module initiates a connection message to the IP phone, IAD, or ONU based on the dial-up information. This message is sent through the Ethernet switching module and the digital Ethernet interface provided by the network address translation module.

[0018] S5: After receiving the SIP message, the IP phone, IAD, or ONU exchanges signals with the dispatching switch through the direction channel of the above process to negotiate and establish a voice channel.

[0019] S6: After the voice channel is established, the voice information of both parties is converted from analog to digital or digital to analog through the voice codec module. The analog signal sent by the dispatching switch is converted into an IP data packet and sent from the digital Ethernet interface to the IP phone, IAD or ONU. Conversely, the message sent by the IP phone, IAD or ONU is sampled and converted into an analog signal and sent from the telephone line interface.

[0020] S7: After the session is established, the voice codec module continuously performs dynamic bandwidth adaptation from 32kbps to 64kbps, and the embedded firewall module performs SIP protocol deep packet inspection in real time;

[0021] S8: When the call ends, either party sends a release request to terminate the session, and the device releases the corresponding codec resources and ports.

[0022] As an improvement, direct telephone call dispatch includes the following steps:

[0023] S1: The direct call initiates a call request. The IP phone, IAD, or ONU sends a connection invitation, which is sent to the direct call protocol converter via the digital Ethernet interface.

[0024] S2: The direct telephone protocol converter forwards the connection invitation to the SIP protocol processing module through the network address translation module and the Ethernet switching module;

[0025] S3: The SIP protocol processing module identifies the invitation message and interacts with the analog signaling processing module based on the message content.

[0026] S4: The analog signaling processing module uses dual-tone multi-frequency signals to establish a call connection with the dispatching switch.

[0027] S5: The dispatching switch communicates with the IP phone, IAD, or ONU through the reverse channel to negotiate and establish a voice channel;

[0028] S6: After the voice channel is established, the voice information of both parties is converted from analog to digital or digital to analog via the voice codec module. The analog signal sent by the dispatching switch is converted into an IP data packet and sent through the digital Ethernet interface to the IP phone, IAD, or ONU. Conversely, the message sent by the IP phone, IAD, or ONU is sampled and converted into an analog signal and sent out through the telephone line interface.

[0029] S7: After the session is established, the voice codec module continuously performs dynamic bandwidth adaptation from 32kbps to 64kbps, and the embedded firewall module performs SIP protocol deep packet inspection in real time;

[0030] S8: When the call ends, either party sends a release request to terminate the session, and the device releases the corresponding codec resources and ports.

[0031] Compared with the existing technology, the advantages of the present invention are: realizing voice calls between GPON / XGS-PON ONU electric direct telephone users in the section booths and the station dispatch desk / duty desk; realizing voice calls between GPON ONU electric direct telephone users in the section optical communication poles and the station dispatch desk / duty desk; making full use of existing digital dispatching switches and railway GPON / XGS-PON systems to truly reduce costs and increase efficiency; realizing stable and reliable electric direct telephone services; meeting the different numbers of electric direct telephone service activation requirements of each station; and realizing service protection by means of link aggregation of the railway GPON / XGS-PON system, shielding all communication ports except voice protocols, and ensuring network security. BRIEF DESCRIPTION OF THE DRAWINGS

[0032] The accompanying drawings are used to provide a further understanding of the technical solution of the present invention and constitute a part of the specification. Together with the embodiments of the present application, they are used to explain the technical solution of the present invention and do not constitute a limitation on the technical solution of the present invention.

[0033] Figure 1 A diagram of a module for a direct-to-phone protocol converter provided in an embodiment of the present application;

[0034] Figure 2 A network topology diagram of the scheduling system provided in an embodiment of the present application;

[0035] Figure 3 Flowcharts provided for embodiments of the present application;

[0036] Figure 4 The process provided for the embodiment of this application Figure 2 ;

[0037] Figure 5 A timing diagram is provided for an embodiment of the present application. DETAILED DESCRIPTION

[0038] The following will be combined with the drawings in the embodiments of this application to clearly and completely describe the technical solutions in the embodiments of this application. Obviously, the embodiments described are only part of the embodiments of this application, not all of the embodiments. Based on the embodiments in this application, all other embodiments obtained by ordinary technicians in this field without making creative efforts are within the scope of protection of this application.

[0039] It should be noted that all directional indications in the embodiments of the present invention (such as up, down, left, right, front, back, etc.) are only used to explain the relative position relationship, movement status, etc. between the various components under a certain specific posture (as shown in the accompanying drawings). If the specific posture changes, the directional indication will also change accordingly.

[0040] The terms "first" and "second" are used for descriptive purposes only and should not be construed as indicating or implying relative importance or implicitly specifying the number of the technical features being referred to. Thus, a feature specified as "first" or "second" may explicitly or implicitly include one or more of such features. Throughout this application, unless otherwise specified, "plurality" means two or more.

[0041] In the description of this application, it should be noted that, unless otherwise expressly specified or limited, the terms "connected" and "connect" should be understood in a broad sense. For example, they can refer to fixed connections, detachable connections, or integral connections. Those skilled in the art will understand the specific meanings of the above terms in this application based on the specific circumstances. Furthermore, when describing pipelines, the terms "connected" and "connected" used in this application have the meaning of conducting electricity. The specific meanings need to be understood in the context.

[0042] In the embodiments of this application, words such as "exemplary" or "for example" are used to indicate examples, illustrations, or descriptions. Any embodiment or design described as "exemplary" or "for example" in the embodiments of this application should not be interpreted as being preferred or advantageous over other embodiments or designs. Rather, the use of words such as "exemplary" or "for example" is intended to present the relevant concepts in a concrete manner.

[0043] like Figure 1 An embodiment of the present application provides a railway electric dispatch direct telephone protocol converter based on analog-to-digital conversion, including seven parts: a network address translation module, an Ethernet switching module, a SIP protocol processing module, a voice codec module, a TDM switching module, an analog signaling processing module, and a Z interface module.

[0044] It has realized the conversion and adaptation of different interfaces and protocols in the railway electric dispatching direct telephone system, provided basic equipment support for the subsequent upgrade and optimization of the railway communication system, achieved the beneficial effect of integrating different communication technologies and realizing the modernization of the railway electric dispatching direct telephone service, and promoted the upgrading of railway communication equipment.

[0045] As an improvement, the analog-to-digital conversion-based direct-telephone protocol converter is provided in the direct-telephone protocol converter dispatching communication system.

[0046] The protocol converter is installed in the dispatching communication system of the direct telephone protocol converter, so that the protocol converter can be integrated into the overall architecture of railway communications, playing its role as a key component of the system and ensuring the integrity and coordination of the railway communication system.

[0047] like Figure 2As an improvement, the dispatching system includes a dispatching office dispatching switch device, which is connected to the station dispatching switch device via an E1 interface. The station dispatching switch is connected to a telephone protocol converter via a telephone line, and the telephone protocol converter is connected to a data network or a PON network (passive optical network) via a digital Ethernet interface. Finally, an IP phone, IAD (integrated access device) or ONU (optical network unit) is connected through the data network or PON network. The IP phone can be used as a direct-through telephone; the IAD or ONU can be connected to an ordinary telephone as a direct-through telephone. The telephone protocol converter is an analog-to-digital conversion-based direct-through telephone protocol converter.

[0048] This technology extends and expands the railway dispatching direct telephone service from traditional dispatching equipment to optical access networks. On the one hand, it fully leverages the large capacity and high speed advantages of modern optical communication technology to improve the transmission quality of railway dispatching direct telephone services. On the other hand, it achieves comprehensive communication coverage of all stations and sections along the railway, improving the accessibility and reliability of railway communications and providing broader and more stable communication support for railway transportation dispatching.

[0049] As an improvement, the dispatching switch of the dispatching office and the station dispatching switch are connected to the dispatching console through telephone lines, and the optical network units of the office booth and the interval optical talk column are connected to the electric dispatch direct telephone through telephone lines.

[0050] This connection method not only retains the original dispatching console's direct control and management functions for telephone communications, but also enables new communication equipment such as section optical communication poles to be effectively connected to the system, realizing the compatibility and unification of railway electric dispatching direct telephone services in traditional and modern communication technology environments.

[0051] As an improvement, the optical line terminal (OLT) and optical network unit (ONU) are based on one of the three standard protocols of GPON, XGPON or XGS-PON.

[0052] GPON and XGS-PON technologies offer advantages such as high bandwidth, long-distance transmission, and excellent optical splitting properties. They can meet the simultaneous access needs of a large number of communication devices along the railway line, providing sufficient data transmission capacity for railway power dispatch direct telephone services and other multimedia communication services. Furthermore, the use of a unified optical access protocol facilitates unified management and maintenance of system equipment, improving the stability and reliability of the communication network and reducing operation and maintenance costs.

[0053] As an improvement, a Gigabit / 100M network connection is adopted between the telephone protocol converter and the station optical line terminal.

[0054] It greatly improves the transmission rate and data throughput of the communication link, and can meet the strict requirements of railway direct telephone services in terms of high bandwidth and low latency.

[0055] As an improvement, a dual / single E1 link is set between the dispatching switch of the dispatching office and the station dispatching switch, and a dual / single link is set between the telephone protocol converter and the station optical line terminal.

[0056] The introduction of a redundant backup mechanism significantly improves the reliability of the railway communication system. If the primary link fails or experiences an anomaly, the backup link automatically and quickly switches to assume communication duties, ensuring uninterrupted railway communication. Furthermore, the dual-link design can also share communication traffic to a certain extent, improving the overall performance and efficiency of the system.

[0057] As an improvement, the feature is that the voice codec module has a built-in programmable DSP chip and supports dynamic switching of three coding formats: G.711, G.729 and AMR-WB.

[0058] This dynamic switching capability optimizes the coding efficiency of voice signals, improving their anti-interference capabilities and compression ratio during transmission, and reducing network bandwidth usage. Furthermore, it ensures voice quality by selecting the appropriate coding format in different communication scenarios, such as short-range and long-range communications, or in weak signal environments, enhancing the user experience. Furthermore, the programmable DSP chip possesses powerful signal processing capabilities, enabling real-time processing and optimization of voice signals, further improving the quality and stability of voice communications. This provides clear and smooth voice calls for railway electric dispatch direct telephone services, meeting the high-fidelity voice communication requirements of railway transportation dispatchers.

[0059] As an improvement, the analog signaling processing module analyzes signals such as off-hook, on-hook, fork flash, and dialing through the dual-tone multi-frequency signal processing module.

[0060] The dual-tone multi-frequency signal recognition module quickly and accurately identifies the DTMF signals generated by dialing, ensuring timely acquisition and correct interpretation of dialing information. The time-division multiplexing switching module rationally multiplexes and distributes the parsed signaling with other communication signals, improving communication link utilization and avoiding interference and conflicts during signaling transmission.

[0061] like Figure 3-4 As an improvement, dispatching the direct call includes the following steps:

[0062] S1: The dispatch office initiates a call request through the dispatch desk / duty desk, which is then sent to the direct telephone protocol converter via the station dispatch switch.

[0063] S2: The direct telephone protocol converter receives analog voice signals through the Z interface and distributes them to the analog signaling processing module through the TDM (time division multiplexing) switching module. The analog signaling processing module analyzes the voice signals in real time and resolves the dialing information therein.

[0064] S3: The analog signaling processing module sends the parsed dialing information to the SIP protocol processing module.

[0065] S4: The SIP protocol processing module initiates a connection message to the IP phone, IAD, or ONU based on the dial-up information. This message is sent through the Ethernet switching module and the digital Ethernet interface provided by the network address translation module.

[0066] S5: After receiving the SIP message, the IP phone, IAD or ONU interacts with the dispatching switch through the direction channel of the above process to negotiate and establish a voice channel.

[0067] S6: After the voice channel is established, the voice information from both sides undergoes analog-to-digital or digital-to-analog conversion via the voice codec module. The analog signal from the dispatching switch is converted into an IP packet and sent through the digital Ethernet interface to the IP phone, IAD, or ONU. Conversely, the packet from the IP phone, IAD, or ONU is sampled and converted into an analog signal and sent through the telephone line interface.

[0068] S7: After the session is established, the voice codec module continuously performs dynamic bandwidth adaptation from 32kbps to 64kbps, and the embedded firewall module performs SIP protocol deep packet inspection in real time;

[0069] S8: When the call ends, either party sends a release request to terminate the session, and the device releases the corresponding codec resources and ports.

[0070] As an improvement, direct telephone call dispatch includes the following steps:

[0071] S1: The direct call initiates a call request. The IP phone, IAD or ONU sends a connection invitation, which is sent to the direct call protocol converter via the digital Ethernet interface.

[0072] S2: The direct telephone protocol converter forwards the connection invitation to the SIP protocol processing module through the network address translation module and the Ethernet switching module.

[0073] S3: The SIP protocol processing module identifies the invitation message and interacts with the analog signaling processing module based on the message content.

[0074] S4: The analog signaling processing module uses dual-tone multi-frequency signals to establish a call connection with the dispatching switch.

[0075] S5: The dispatching switch communicates with the IP phone, IAD, or ONU through the reverse channel to negotiate and establish a voice channel.

[0076] S6: After the voice channel is established, the voice information from both sides undergoes analog-to-digital or digital-to-analog conversion via the voice codec module. The analog signal from the dispatching switch is converted into an IP packet and sent through the digital Ethernet interface to the IP phone, IAD, or ONU. Conversely, the packet from the IP phone, IAD, or ONU is sampled and converted into an analog signal and sent through the telephone line interface.

[0077] S7: After the session is established, the voice codec module continuously performs dynamic bandwidth adaptation from 32kbps to 64kbps, and the embedded firewall module performs SIP protocol deep packet inspection in real time.

[0078] S8: When the call ends, either party sends a release request to terminate the session, and the device releases the corresponding codec resources and ports.

[0079] Each step is interconnected and synergistic, fully leveraging the functional characteristics of components such as protocol converters, optical line terminals, and digital modulation equipment. This enables railway telephony direct communication to achieve key operations such as precise conversion of analog and digital signals, optimized encoding and decoding of voice signals, accurate identification and transmission of signaling, and effective conversion and management of network addresses. Furthermore, the introduction of dynamic bandwidth adaptation and deep packet inspection (DPI) within the communication process further enhances system performance and security.

[0080] 1. Hardware device model

[0081] GPON / XGS-PON Optical Line Terminal (OLT)

[0082] Model: ZTE ZXA10-C69E (supports GPON / XGS-PON hybrid access).

[0083] Features: Supports Combo mode and is compatible with hybrid access of three types of ONUs: GPON, XG-PON, and XGS-PON.

[0084] Voice codec module DSP chip

[0085] Model: Huawei MA5683T (supports GPON / XGS-PON hybrid access).

[0086] Features: Supports Combo mode and is compatible with hybrid access of three types of ONUs: GPON, XG-PON, and XGS-PON.

[0087] Voice codec module DSP chip

[0088] Model: TI TMS320C6748 (supports G.711, G.729, and AMR-WB dynamic switching).

[0089] Features: 32-bit floating-point unit, supports real-time voice processing.

[0090] Gigabit network switching module

[0091] Model: MikroTik CRS326-24G-2S+RM (24-port Gigabit switch).

[0092] Features: Supports link aggregation (LACP) and redundant backup.

[0093] Railway digital walkie-talkie

[0094] Model: ZTE Gundam RH506.

[0095] Features: Supports AMBE++ vocoder, suitable for voice communication in complex environments.

[0096] 2. Existing Software and Protocols

[0097] SIP protocol stack

[0098] Software name: PJSIP (open source SIP protocol stack).

[0099] Function: Support RFC 3261 standard for session establishment and termination (see claims 10S4, S8).

[0100] Network Address Translation (NAT) module

[0101] Software name: Linux Netfilter / NAT (kernel-level implementation).

[0102] Function: Dynamic port mapping (16384-32768), only open SIP (5060) and RTP ports (see claim 10S5).

[0103] Speech Codec Library

[0104] G.711: Open source implementation (such as GStreamer).

[0105] G.729: Cisco-licensed implementation of G.729A.

[0106] AMR-WB: 3GPP standard open source library (such as OpenCORE).

[0107] Dual-tone multi-frequency (DTMF) recognition algorithm

[0108] Software name: Goertzel algorithm (open source mathematical library).

[0109] Function: Real-time detection of DTMF signal frequency.

[0110] Protocol interaction process

[0111] SIP protocol: Session establishment (INVITE), maintenance (200OK) and termination (BYE) are implemented through the PJSIP library.

[0112] RTP protocol: encapsulates voice data packets (RTP / AVP) and supports dynamic encoding format switching.

[0113] Network security design

[0114] NAT module: Only allows SIP (5060) and RTP (16384-32768) port communications, blocking other ports (such as TCP80 / 443).

[0115] Firewall module: Detects illegal signaling (such as unauthorized REGISTER requests) in real time through the SIP DPI algorithm.

[0116] Redundant backup mechanism

[0117] Automatic dual-link switching: Through link status monitoring (such as ping detection), the backup link is automatically enabled when the main link fails.

[0118] The above is only a specific embodiment of the present application, but the scope of protection of the present application is not limited thereto. Any changes or replacements within the technical scope disclosed in the present application should be included in the scope of protection of the present application. Therefore, the scope of protection of the present application should be based on the scope of protection of the claims.

Claims

1. A railway electric speed control direct telephone protocol converter based on analog-to-digital conversion, characterized in that: include: Network address translation module, Ethernet switching module, SIP protocol processing module, voice codec module, TDM switching module, analog signaling processing module, Z interface module, the network address translation module has a built-in digital Ethernet interface, the Ethernet switching module is connected to the SIP protocol processing module, the voice codec module and the network address translation module, the TDM switching module is connected to the Z interface module, the voice codec module and the analog signaling processing module, and the analog signaling processing module interacts with the SIP protocol processing module.

2. The analog-to-digital conversion-based telephony protocol converter according to claim 1, characterized in that: The analog-to-digital conversion-based electric dispatching direct telephone protocol converter is arranged in a railway dispatching communication system.

3. The analog-to-digital conversion-based telephony protocol converter according to claim 1, characterized in that: The electric dispatching direct telephone protocol converter is connected to the railway digital dispatching switch through a telephone line, connected to the data network through a digital Ethernet interface, and connected to one of the three devices of IP telephone, IAD and ONU through the data network.

4. The railway electric speed control direct telephone protocol converter based on analog-to-digital conversion according to claim 3 is characterized in that: The PON network consists of an OLT, an optical splitter, and an ONU, and is based on one of the three standard protocols: GPON, XGPON, or XGS-PON.

5. The dispatching system according to claim 3, characterized in that: The telephone protocol converter and the IP phone, IAD and ONU are connected using a 100M network.

6. The analog-to-digital conversion-based telephony protocol converter according to claim 1, characterized in that: The voice codec module has a built-in programmable DSP chip and supports dynamic switching of three coding formats: G.711, G.729 and AMR-WB.

7. The analog-to-digital conversion-based telephony protocol converter according to claim 1, characterized in that: The analog signaling processing module can detect or generate DTMF analog signaling signals.

8. The analog-to-digital conversion-based telephony protocol converter according to claim 1, characterized in that: The SIP protocol is used between IP phones, IADs, and ONUs.

9. The working method of the railway electric speed control direct telephone protocol converter based on analog-to-digital conversion according to claim 3 is characterized in that: The dispatch call involves the following steps: S1: The dispatch office initiates a call request through the dispatch desk / duty desk, which is then sent to the direct telephone protocol converter via the station dispatch switch. S2: The direct telephone protocol converter receives analog voice signals through the Z interface and distributes them to the analog signaling processing module through the TDM switching module. The analog signaling processing module analyzes the voice signals in real time and parses the dialing information. S3: The analog signaling processing module sends the parsed dialing information to the SIP protocol processing module; S4: The SIP protocol processing module initiates a connection message to the connected device based on the dial-up information. The message is sent from the digital Ethernet interface provided by the network address translation module through the Ethernet switching module. S5: After receiving the SIP message, the connected device interacts with the dispatching switch through the direction channel of the above process to negotiate and establish a voice channel; S6: After the voice channel is established, the voice information of both parties is converted from analog to digital or digital to analog through the voice codec module. The analog signal sent by the dispatching switch is converted into an IP data packet and sent to the connected device through the digital Ethernet interface. Conversely, the message sent by the connected device is sampled and converted into an analog signal and sent out through the telephone line interface. S7: After the session is established, the voice codec module continuously performs dynamic bandwidth adaptation from 32kbps to 64kbps, and the embedded firewall module performs SIP protocol deep packet inspection in real time; S8: When the call ends, either party sends a release request to terminate the session, and the device releases the corresponding codec resources and ports.

10. The working method of the railway electric speed control direct telephone protocol converter based on analog-to-digital conversion according to claim 3, characterized in that: Direct call to the dispatch office The following steps are involved: S1: The direct call initiates a call request, and the connected device sends a connection invitation, which is sent to the direct call protocol converter via the digital Ethernet interface; S2: The direct telephone protocol converter forwards the connection invitation to the SIP protocol processing module through the network address translation module and the Ethernet switching module; S3: The SIP protocol processing module identifies the invitation message and interacts with the analog signaling processing module based on the message content; S4: The analog signaling processing module uses dual-tone multi-frequency signals to establish a call connection with the dispatching switch; S5: The dispatching switch negotiates with the connected device through the reverse channel to establish a voice channel; S6: After the voice channel is established, the voice information of both parties is converted from analog to digital or digital to analog through the voice codec module. The analog signal sent by the dispatching switch is converted into an IP data packet and sent to the connected device through the digital Ethernet interface. Conversely, the message sent by the connected device is sampled and converted into an analog signal and sent out through the telephone line interface. S7: After the session is established, the voice codec module continuously performs dynamic bandwidth adaptation from 32kbps to 64kbps, and the embedded firewall module performs SIP protocol deep packet inspection in real time; S8: When the call ends, either party sends a release request to terminate the session, and the device releases the corresponding codec resources and ports.

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