CN110971582A - Center-level voice interconnection and intercommunication method and system between 800M and 350M systems - Google Patents
Center-level voice interconnection and intercommunication method and system between 800M and 350M systems Download PDFInfo
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1101—Session protocols
- H04L65/1104—Session initiation protocol [SIP]
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L63/00—Network architectures or network communication protocols for network security
- H04L63/08—Network architectures or network communication protocols for network security for authentication of entities
- H04L63/0815—Network architectures or network communication protocols for network security for authentication of entities providing single-sign-on or federations
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/10—Architectures or entities
- H04L65/102—Gateways
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1083—In-session procedures
- H04L65/1086—In-session procedures session scope modification
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/40—Support for services or applications
- H04L65/4061—Push-to services, e.g. push-to-talk or push-to-video
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
- H04L65/61—Network streaming of media packets for supporting one-way streaming services, e.g. Internet radio
- H04L65/611—Network streaming of media packets for supporting one-way streaming services, e.g. Internet radio for multicast or broadcast
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
- H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
Abstract
The invention relates to a method and a system for center-level voice interconnection and intercommunication between 800M and 350M systems, wherein the method comprises the following steps: step S1: reading SIP configuration and configuration information and system information of an 800M system and a 350M system; step S2: establishing a SIP session according to the pre-configured configuration information, and establishing communication with the 800M digital trunking communication system and communication with the 350M analog trunking communication system; step S3: if the incoming call information of the 800M digital trunking communication system or the 350M analog trunking communication system is detected, then performing step S4, if the incoming call information of the SIP client is detected, performing step S7; step S4: if the call type is incoming call, executing step S5, and if the call type is call end, executing step S6; step S5: transmitting a voice stream; step S6: and terminating the transmission of the voice stream; step S7: and receiving the SIP Message and the voice stream of the opposite terminal, and transmitting the SIP Message and the voice stream to a corresponding communication system. Compared with the prior art, the method and the device have the advantages of flexible configuration, strong expansibility and the like.
Description
Technical Field
The invention relates to a voice scheduling technology, in particular to a method and a system for central-level voice interconnection and intercommunication between 800M and 350M systems.
Background
There are currently two sets of voice communication systems in some systems, including, for example, 350M analog trunking communication systems provided by some manufacturers and 800M digital trunking communication systems provided by european space. The 350M system is established earlier, the whole communication of an early government affair network is realized, and with the increase of technology and the improvement of demand, an 800M digital trunking communication system is adopted in the subsequent network construction. In order to realize the voice intercommunication between the 800M digital trunking communication system and the reserved 350M analog trunking communication system and ensure that police and security personnel in the dispatching places of public security in the subway can receive dispatching commands, the interconnection and intercommunication between the two different systems in the same line network must be realized.
Disclosure of Invention
The present invention is directed to a method and system for center-level voice interworking between 800M and 350M systems to overcome the above-mentioned drawbacks of the prior art.
The purpose of the invention can be realized by the following technical scheme:
a method for center-level voice interconnection and intercommunication between 800M and 350M systems comprises the following steps:
step S1: reading SIP configuration and configuration information and system information of an 800M digital trunking communication system and a 350M analog trunking communication system;
step S2: establishing a SIP session according to the pre-configured configuration information, and establishing communication with the 800M digital trunking communication system and communication with the 350M analog trunking communication system;
step S3: simultaneously detecting whether a call exists in the 800M digital trunking communication system and the 350M analog trunking communication system and incoming call information of the SIP client, if the incoming call information of the 800M digital trunking communication system or the 350M analog trunking communication system is detected, then executing step S4, and if the incoming call information of the SIP client is detected, executing step S7;
step S4: judging the call type, if the call type is an incoming call, executing the step S5, and if the call type is an end of call, executing the step S6;
step S5: initiating a PTT to press down an SIP Message by using a pre-bound SIP client, and transmitting a voice stream;
step S6: initiating the SIP Message released by the PTT by using the pre-bound SIP client, and terminating the transmission of the voice stream;
step S7: and receiving the SIP Message and the voice stream of the opposite terminal, and transmitting the SIP Message and the voice stream to a corresponding communication system.
The step S7 specifically includes:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the type is 800M digital trunking communication system, executing step S74, if the type is 350M analog trunking communication system, executing step S73;
step S73: transcoding and sending the voice stream;
step S74: and according to the SIP Message, an 800M talk group is appointed on an appointed TCS client to carry out PTTup and PTT down operations, and the taken voice stream is transcoded and sent.
The judgment criterion in the step S4 specifically includes:
for a call from an 800M digital trunking communication system, judging the type of the call according to the event mode of call information;
for a call from a 350M analog trunked communication system, the first receipt of a voice stream is considered an originating call and the end of the call is considered if the voice stream is no longer received within 100 milliseconds.
The step S2 specifically includes:
step S21: establishing an SIP session according to the pre-configured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice stream on a designated IP channel;
step S23: and establishing communication through a private protocol provided by a 350M system, logging in the 350M gateway by adopting the allocated agent number, and loading the talk group on the designated gateway.
The method further comprises the following steps:
and the SIP ID allocated to each physical link establishes a group calling configuration relation table according to the interconnected and intercommunicated call group ID and the physical link associated with the call group ID, and the group calling SIP gateway acquires the group calling configuration relation table after being started.
A center-level voice interconnection and interworking system between 800M and 350M systems, comprising an 800M digital trunking communication system and a 350M analog trunking communication system, and further comprising a control server connected to the 800M digital trunking communication system and the 350M analog trunking communication system, respectively, the control server comprising a memory and a processor, and a program stored in the memory and executed by the processor, the processor implementing the following steps when executing the program:
step S1: reading SIP configuration and configuration information and system information of an 800M digital trunking communication system and a 350M analog trunking communication system;
step S2: establishing a SIP session according to the pre-configured configuration information, and establishing communication with the 800M digital trunking communication system and communication with the 350M analog trunking communication system;
step S3: simultaneously detecting whether a call exists in the 800M digital trunking communication system and the 350M analog trunking communication system and incoming call information of the SIP client, if the incoming call information of the 800M digital trunking communication system or the 350M analog trunking communication system is detected, then executing step S4, and if the incoming call information of the SIP client is detected, executing step S7;
step S4: judging the call type, if the call type is an incoming call, executing the step S5, and if the call type is an end of call, executing the step S6;
step S5: initiating a PTT to press down an SIP Message by using a pre-bound SIP client, and transmitting a voice stream;
step S6: initiating the SIP Message released by the PTT by using the pre-bound SIP client, and terminating the transmission of the voice stream;
step S7: and receiving the SIP Message and the voice stream of the opposite terminal, and transmitting the SIP Message and the voice stream to a corresponding communication system.
The step S7 specifically includes:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the communication system is an 800M digital trunking communication system, executing step S74, and if the communication system is a 350M analog trunking communication system, executing step S73;
step S73: transcoding and sending the voice stream;
step S74: and according to the SIP Message, an 800M talk group is appointed on an appointed TCS client to carry out PTTup and PTT down operations, and the taken voice stream is transcoded and sent.
The judgment criterion in the step S4 specifically includes:
for a call from an 800M digital trunking communication system, judging the type of the call according to the event mode of call information;
for a call from a 350M analog trunked communication system, the first receipt of a voice stream is considered an originating call and the end of the call is considered if the voice stream is no longer received within 100 milliseconds.
The step S2 specifically includes:
step S21: establishing an SIP session according to the pre-configured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice stream on a designated IP channel;
step S23: and establishing communication through a private protocol provided by a 350M system, logging in the 350M gateway by adopting the allocated agent number, and loading the talk group on the designated gateway.
The processor, when executing the program, further implements the steps of:
and the SIP ID allocated to each physical link establishes a group calling configuration relation table according to the interconnected and intercommunicated call group ID and the physical link associated with the call group ID, and the group calling SIP gateway acquires the group calling configuration relation table after being started.
Compared with the prior art, the invention has the following beneficial effects:
1) the scheme adopts the SIP mode, thus supporting distributed construction, and having flexible configuration and strong expansibility.
2) For the system side, the system is equivalent to a dispatching desk which simultaneously realizes two system platforms. The voice link is converted into a local sound card device and a microphone, and the local sound card device and the microphone can be used as a dispatching desk.
3) The final realization adopts a centralized mode, for an 800M system, the mode realizes the simultaneous login of a plurality of TCS clients, for the original 800M system dispatching desk, the original dispatching desk is a device for logging in a number, at present, the mode proves that a plurality of numbers can be used for logging in simultaneously, a plurality of TCS clients can carry out operations such as calling simultaneously, and other services can be realized by providing another way for the 800M dispatching system later.
4) The method provides some technical points for the digital processing (coding and decoding and transcoding) of the voice stream.
5) And the nodes which are interconnected and intercommunicated are flexibly selected.
Drawings
FIG. 1 is a schematic flow chart of the main steps of the method of the present invention;
FIG. 2 is a block diagram of the system of the present invention;
wherein: 1. 350M analog trunking communication system, 2, 800M digital trunking communication system, 3, control server.
Detailed Description
The invention is described in detail below with reference to the figures and specific embodiments. The present embodiment is implemented on the premise of the technical solution of the present invention, and a detailed implementation manner and a specific operation process are given, but the scope of the present invention is not limited to the following embodiments.
The system adopts a central-level digital interconnection mode, and firstly, the system can communicate in a digital mode. For the 800M system side, the SDK is provided by using the system, the signaling communication with the TCS server is established in a scheduling mode, and the voice communication with the 800M system TVG is established through an RTP protocol, so that a path of group calling function can be established. For the 350M system side, a Vide self-defined private protocol is adopted to establish communication with the 350M system gateway, voice communication is carried out through an RTP protocol, and a path of group calling is established. Then, by adopting SIP session protocol, one path of interconnection and intercommunication of 800M-350M system can be completed by using a pair of SIP clients.
The design of the application adopts a distributed scheme, namely one device establishes communication with one end system and one path of SIP client, but in order to save cost, the system side also supports a mode of completing multi-path simultaneous communication by one device, and for an 800M system, a mode of logging in a plurality of TCS clients simultaneously on one device can be adopted to realize multi-path simultaneous group calling function. For the 350M side, the system gateway supports a multi-path simultaneous group calling function. Therefore, one device can be used for establishing communication with two-end systems and simultaneously using multiple SIP clients to complete multiple simultaneous interconnection and interworking. There is provided a center-level voice interconnection and interworking system between 800M and 350M systems, which includes an 800M digital trunking communication system 2 and a 350M analog trunking communication system 1 as shown in fig. 2, and further includes a control server 3 connected to the 800M digital trunking communication system 2 and the 350M analog trunking communication system 1, respectively, as shown in fig. 1, the control method adopted by the control server includes:
firstly, the SIP ID allocated to each physical link establishes a group calling configuration relation table according to the interconnected and intercommunicated talk group ID and the physical link associated with the talk group ID, and the group calling SIP gateway acquires the group calling configuration relation table after being started.
Step S1: reading SIP configuration and configuration information and system information of an 800M digital trunking communication system 2 and a 350M analog trunking communication system 1; establishing different group calling configuration relation tables for each group calling SIP gateway for the conversation group ID, the physical link associated with the conversation group ID and the SIP ID distributed by the physical link which are required to be interconnected and intercommunicated by the two systems, and automatically establishing dial-up connection between the corresponding SIP IDs through an SIP exchange server according to the obtained group calling configuration relation tables after a user logs in the group calling SIP gateway; the administrator can flexibly modify the group call configuration relation table. Specifically, the authorized user modifies the configuration file, establishes a one-to-one group call configuration relationship for the talk group, the physical port number associated with the talk group ID, and the SIP ID, and also needs to configure an IP address and a port number of an SIP exchange server, TCS server information of the 800M system, a login user name, a login password, and a group call number, and gateway information of the 350M system includes an agent number, a gateway IP, and the like.
Step S2: establishing a SIP session according to the preconfigured configuration information, and establishing communication with the 800M digital trunking communication system 2 and communication with the 350M analog trunking communication system 1, specifically comprising:
step S21: establishing an SIP session according to the pre-configured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice stream on a designated IP channel;
step S23: and establishing communication through a private protocol provided by a 350M system, logging in the 350M gateway by adopting the allocated agent number, and loading the talk group on the designated gateway.
Establishing communication with an 800M system, including simultaneously logging in by using a plurality of accounts through an SDK provided by the system, establishing a communication flow with a TCS server of the 800M system, establishing a plurality of TCS clients, and selecting a group number to be used as interconnection in each TCS client to be loaded as a main resource. Establishing communication with the 350M system includes using an internal protocol provided by the system, logging in to and establishing communication with the 350M gateway using the agent number assigned by the system, and then loading the required interconnected group number (also referred to as the so-called number at the 350M system) on the corresponding gateway. Namely: after the interconnection server is started, an SIP session is established according to the information of the configuration table, and two SIP numbers are paired; then, communication with the 800M system is established through the SDK provided by the 800M system, the information of the using talk group is loaded, and meanwhile, voice streams are monitored on a designated IP channel; and finally, establishing communication through a private protocol provided by a 350M system, logging in the 350M gateway by adopting the allocated agent number, and loading the talk group on the designated gateway.
Step S3: simultaneously detecting whether a call exists in the 800M digital trunking communication system 2 and the 350M analog trunking communication system 1 and the incoming call information of the SIP client, if detecting the incoming call information of the 800M digital trunking communication system 2 or the 350M analog trunking communication system 1, then executing step S4, if detecting the incoming call information of the SIP client, executing step S7;
the device detects that a certain TCS client of the 800M system has an incoming call signaling and simultaneously takes a corresponding voice stream through an RTP (real-time transport protocol) protocol, because no additional information is needed for a 350M incoming call (such as a PTT (push-to-talk) push-to-talk state and the like), the voice stream is directly transcoded and directly sent to an opposite-end SIP client through the corresponding SIP client, the opposite-end SIP client transcodes the voice stream (to a code stream required by the 350M system) after obtaining the voice, and then the voice stream is sent to the 350M voice gateway through the RTP protocol of the end, so that the function of answering the 350M system pushed out from the 800M system is realized.
Step S4: judging the call type, if the call type is an incoming call, executing step S5, and if the call type is an end of call, executing step S6, wherein the judgment basis specifically includes:
for a call from the 800M digital trunking communication system 2, judging the type of the call according to the event mode of the call information;
for a call from the 350M analog trunked communication system 1, the first receipt of a voice stream is considered an originating call and the end of the call is considered if the voice stream is no longer received within 100 milliseconds.
That is, in this step, whether the call information is incoming call information or call end information is detected, and for the 800M system, which TCS client is the call information can be detected according to the event mode of the SDK; for a 350M system, detection is based on voice stream, and if the voice stream is received for the first time, the opposite-end originating call is considered, and if the voice stream is not received within 100 milliseconds, the opposite-end call is considered to be ended.
Step S5: initiating a PTT to press down an SIP Message by using a pre-bound SIP client, and transmitting a voice stream;
step S6: initiating the SIP Message released by the PTT by using the pre-bound SIP client, and terminating the transmission of the voice stream;
when detecting that a certain group of a 350M system has an incoming call (by detecting that voice data exists in a corresponding RTP protocol), the device firstly sends an SIP message indicating that an opposite terminal PTT is pressed down to an opposite terminal SIP client through the SIP client, and after receiving the message, the SIP client presses down the PTT of the loaded group on a corresponding TCS client (the device is communicated with a TCS server of an 800M system); then, the SIP client at the 350M system side transcodes the obtained voice stream and sends the voice stream, and the SIP client at the opposite end transcodes the received voice stream and sends the voice stream to the TVG server through RTP. Thus, the function of answering from the 350M outgoing call 800M system is realized. When no voice is detected for more than 100 milliseconds, a SIP message for the opposite party PTT release is sent through the SIP client. The opposite SIP client detects the message and releases the PTT on the corresponding TCS client.
Step S7: receiving an opposite terminal SIP Message and a voice stream, and transmitting the opposite terminal SIP Message and the voice stream to a corresponding communication system, specifically comprising:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the type is 800M digital trunking communication system 2, executing step S74, and if the type is 350M analog trunking communication system 1, executing step S73;
step S73: transcoding and sending the voice stream;
step S74: and according to the SIP Message, an 800M talk group is appointed on an appointed TCS client to carry out PTTup and PTT down operations (through the SDK, and the taken voice stream is transcoded and sent.
The system has a dynamic selection function, namely the SIP client provides a channel, and the 800M system and the 350M system are respectively bound to the corresponding clients, so that the 800M system can be selected to the group which needs to be interconnected when no call exists, and the 350M system can also be selected to the group which needs to be interconnected. Meanwhile, the SIP client can also be manually operated to select the dialed opposite-end SIP number, so that the internal group intercommunication of the same system can be realized.
In addition, the method and the device fully disclose in the application can be realized through pure software improvement and can be realized through software and hardware improvement, for example, a voice board card is configured, and the type of the voice board card can adopt east-in K161A-E.
Claims (10)
1. A method for center-level voice interconnection and interworking between 800M and 350M systems, comprising:
step S1: reading SIP configuration and configuration information, and system information of an 800M digital trunking communication system (2) and a 350M analog trunking communication system (1);
step S2: establishing a SIP session according to the preconfigured configuration information and establishing communication with the 800M digital trunked communication system (2) and communication with the 350M analog trunked communication system (1);
step S3: simultaneously detecting whether a call exists in the 800M digital trunking communication system (2) and the 350M analog trunking communication system (1) and incoming call information of the SIP client, if detecting the incoming call information of the 800M digital trunking communication system (2) or the 350M analog trunking communication system (1), then executing step S4, and if detecting the incoming call information of the SIP client, executing step S7;
step S4: judging the call type, if the call type is an incoming call, executing the step S5, and if the call type is an end of call, executing the step S6;
step S5: initiating a PTT to press down an SIP Message by using a pre-bound SIP client, and transmitting a voice stream;
step S6: initiating the SIP Message released by the PTT by using the pre-bound SIP client, and terminating the transmission of the voice stream;
step S7: and receiving the SIP Message and the voice stream of the opposite terminal, and transmitting the SIP Message and the voice stream to a corresponding communication system.
2. The method for center-level voice interworking between 800M and 350M systems according to claim 1, wherein the step S7 specifically includes:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the type is 800M digital trunking communication system (2), executing step S74, and if the type is 350M analog trunking communication system (1), executing step S73;
step S73: transcoding and sending the voice stream;
step S74: and according to the SIP Message, an 800M talk group is appointed on an appointed TCS client to carry out PTT up and PTTdown operations, and the taken voice stream is transcoded and sent.
3. The method of claim 1, wherein the determining in step S4 specifically includes:
for a call from an 800M digital trunking communication system (2), judging the type of the call according to the event mode of call information;
for a call from a 350M analog trunked communication system (1), the first receipt of a voice stream is considered an originating call and the end of the call is considered if the voice stream is no longer received within 100 milliseconds.
4. The method for center-level voice interworking between 800M and 350M systems according to claim 1, wherein the step S2 specifically includes:
step S21: establishing an SIP session according to the pre-configured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice stream on a designated IP channel;
step S23: and establishing communication through a private protocol provided by a 350M system, logging in the 350M gateway by adopting the allocated agent number, and loading the talk group on the designated gateway.
5. The method of claim 1, wherein the method further comprises:
and the SIP ID allocated to each physical link establishes a group calling configuration relation table according to the interconnected and intercommunicated call group ID and the physical link associated with the call group ID, and the group calling SIP gateway acquires the group calling configuration relation table after being started.
6. A center-level voice interworking system between 800M and 350M systems, comprising an 800M digital trunking communication system (2) and a 350M analog trunking communication system (1), and further comprising a control server (3) connected to the 800M digital trunking communication system (2) and the 350M analog trunking communication system (1), respectively, the control server (3) comprising a memory and a processor, and a program stored in the memory and executed by the processor, the processor implementing the following steps when executing the program:
step S1: reading SIP configuration and configuration information, and system information of an 800M digital trunking communication system (2) and a 350M analog trunking communication system (1);
step S2: establishing a SIP session according to the preconfigured configuration information and establishing communication with the 800M digital trunked communication system (2) and communication with the 350M analog trunked communication system (1);
step S3: simultaneously detecting whether a call exists in the 800M digital trunking communication system (2) and the 350M analog trunking communication system (1) and incoming call information of the SIP client, if detecting the incoming call information of the 800M digital trunking communication system (2) or the 350M analog trunking communication system (1), then executing step S4, and if detecting the incoming call information of the SIP client, executing step S7;
step S4: judging the call type, if the call type is an incoming call, executing the step S5, and if the call type is an end of call, executing the step S6;
step S5: initiating a PTT to press down an SIP Message by using a pre-bound SIP client, and transmitting a voice stream;
step S6: initiating the SIP Message released by the PTT by using the pre-bound SIP client, and terminating the transmission of the voice stream;
step S7: and receiving the SIP Message and the voice stream of the opposite terminal, and transmitting the SIP Message and the voice stream to a corresponding communication system.
7. The system according to claim 6, wherein the step S7 specifically includes:
step S71: receiving an opposite terminal SIP Message and a voice stream;
step S72: judging the type of the communication system pre-bound by the SIP client, if the type is 800M digital trunking communication system (2), executing step S74, and if the type is 350M analog trunking communication system (1), executing step S73;
step S73: transcoding and sending the voice stream;
step S74: and according to the SIP Message, an 800M talk group is appointed on an appointed TCS client to carry out PTT up and PTTdown operations, and the taken voice stream is transcoded and sent.
8. The system according to claim 6, wherein the determining in step S4 specifically includes:
for a call from an 800M digital trunking communication system (2), judging the type of the call according to the event mode of call information;
for a call from a 350M analog trunked communication system (1), the first receipt of a voice stream is considered an originating call and the end of the call is considered if the voice stream is no longer received within 100 milliseconds.
9. The system according to claim 6, wherein the step S2 specifically includes:
step S21: establishing an SIP session according to the pre-configured configuration information;
step S22: establishing communication with the 800M system through the SDK provided by the 800M system, loading and using talk group information, and monitoring voice stream on a designated IP channel;
step S23: and establishing communication through a private protocol provided by a 350M system, logging in the 350M gateway by adopting the allocated agent number, and loading the talk group on the designated gateway.
10. The system of claim 1, wherein the processor, when executing the program, further performs the steps of:
and the SIP ID allocated to each physical link establishes a group calling configuration relation table according to the interconnected and intercommunicated call group ID and the physical link associated with the call group ID, and the group calling SIP gateway acquires the group calling configuration relation table after being started.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201811162064.6A CN110971582B (en) | 2018-09-30 | 2018-09-30 | Central level voice interconnection method and system between 800M and 350M systems |
Applications Claiming Priority (1)
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| CN1585520A (en) * | 2004-06-04 | 2005-02-23 | 中兴通讯股份有限公司 | Interconnecting and interflowing method for digital cluster system and common telephone system |
| CN103200532A (en) * | 2013-04-12 | 2013-07-10 | 哈尔滨海能达科技有限公司 | Device, system and method for achieving interconnection of cluster systems with different patterns |
| CN206602543U (en) * | 2017-03-30 | 2017-10-31 | 通号通信信息集团上海有限公司 | For realizing that different manufacturers TETRA system centres level interconnects system |
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| Publication number | Priority date | Publication date | Assignee | Title |
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| CN1585520A (en) * | 2004-06-04 | 2005-02-23 | 中兴通讯股份有限公司 | Interconnecting and interflowing method for digital cluster system and common telephone system |
| CN103200532A (en) * | 2013-04-12 | 2013-07-10 | 哈尔滨海能达科技有限公司 | Device, system and method for achieving interconnection of cluster systems with different patterns |
| CN206602543U (en) * | 2017-03-30 | 2017-10-31 | 通号通信信息集团上海有限公司 | For realizing that different manufacturers TETRA system centres level interconnects system |
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