CN106302073B - Method and device for realizing call - Google Patents

Method and device for realizing call Download PDF

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Publication number
CN106302073B
CN106302073B CN201510317819.5A CN201510317819A CN106302073B CN 106302073 B CN106302073 B CN 106302073B CN 201510317819 A CN201510317819 A CN 201510317819A CN 106302073 B CN106302073 B CN 106302073B
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voice
data packet
control board
main control
user
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CN106302073A (en
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孙小伟
蓝雅燕
徐劲松
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ZTE Corp
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ZTE Corp
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Priority to CN201510317819.5A priority Critical patent/CN106302073B/en
Priority to RU2017145917A priority patent/RU2687036C1/en
Priority to PCT/CN2016/082339 priority patent/WO2016197796A1/en
Priority to MYPI2017001822A priority patent/MY191001A/en
Publication of CN106302073A publication Critical patent/CN106302073A/en
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L41/00—Arrangements for maintenance, administration or management of data switching networks, e.g. of packet switching networks
    • H04L41/50—Network service management, e.g. ensuring proper service fulfilment according to agreements
    • H04L41/508—Network service management, e.g. ensuring proper service fulfilment according to agreements based on type of value added network service under agreement
    • H04L41/5087—Network service management, e.g. ensuring proper service fulfilment according to agreements based on type of value added network service under agreement wherein the managed service relates to voice services
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00—Data switching networks
    • H04L12/28—Data switching networks characterised by path configuration, e.g. LAN [Local Area Networks] or WAN [Wide Area Networks]
    • H04L12/46—Interconnection of networks
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L9/00—Cryptographic mechanisms or cryptographic arrangements for secret or secure communications; Network security protocols
    • H04L9/40—Network security protocols

Abstract

The invention discloses a method and a device for realizing calling, comprising the following steps: the uplink card receives a resource allocation request from the soft switch equipment and sends the resource allocation request to the main control board; the main control board judges that the real-time transmission protocol RTP coding and decoding resources which can be used exist in the POTS of the voice user board corresponding to the user, distributes first RTP coding and decoding resources for the user according to the RTP coding and decoding resources which can be used on the POTS corresponding to the user, and sends the distributed first RTP coding and decoding resources to the POTS corresponding to the user. By the scheme of the invention, when the POTS corresponding to the user has available RTP coding and decoding resources, the RTP coding and decoding resources available on the POTS corresponding to the user are allocated to the user, so that the probability of call failure of the user is reduced, and the experience of the user is improved.

Description

Method and device for realizing call
Technical Field
The present invention relates to a Voice Over Internet Protocol (VOIP) technology, and more particularly, to a method and apparatus for implementing a call.
Background
A voice subscriber board (POTS) is a subscriber interface unit for narrowband voice access to Internet Protocol (IP), a fixed Telephone in a subscriber's home is connected to the POTS through a copper wire, and the POTS is responsible for converting a voice analog signal from the fixed Telephone into a voice digital signal.
With the development of technology, from the traditional Public Switched Telephone Network (PSTN) mainly based on Telephone switching, a Next Generation Network (NGN) mainly based on packet switching is gradually moved, the NGN carries all services of the original PSTN Network, a large amount of data transmission is offloaded to the IP Network to reduce the load of the PSTN Network, and many new and old services are added and enhanced by the new characteristics of the IP technology.
In short, VOIP digitizes analog signals and transmits them in real time over IP networks in the form of data packets.
Fig. 1 is a schematic structural diagram of a conventional device for implementing a call. As shown in fig. 1, the conventional method for implementing a call roughly includes:
the uplink card receives a resource allocation request from the soft switch equipment and sends the resource allocation request to the main control board; the IP exchange chip of the main control board sends the resource allocation request to the VOIP chip, and the VOIP chip judges that the usable Real-time Transport Protocol (RTP) coding and decoding resources exist in the VOIP chip and allocates the RTP coding and decoding resources for the user.
When the VOIP chip judges that no usable RTP coding and decoding resources are available, the TDM switching chip sends a call failure message to the TDM switching chip, and the TDM switching chip sends the call failure message to the POTS.
After the call is successful, the POTS converts a first voice analog signal from a user into a first voice digital signal and sends the first voice digital signal to the main control board; a Time Division Multiplexing (TDM) exchange chip of the main control board sends the first voice digital signal to a VOIP chip; the VOIP chip adopts the distributed RTP coding and decoding resources to code the first voice digital signal, packages an IP address and a port on the coded first voice digital signal to obtain a first IP voice data packet, and sends the first IP voice data packet to the IP exchange chip; and the IP exchange chip sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
When the uplink card receives the second IP voice data packet, the second IP voice data packet is sent to an IP exchange chip of the main control board; the IP exchange chip sends the second IP voice data packet to the VOIP chip; after the VOIP chip decapsulates the second IP data packet, the decapsulated second IP data packet is decoded by adopting allocated RTP (real-time transport protocol) coding and decoding resources to obtain a second voice digital signal, and the second voice digital signal is sent to the TDM switching chip; the TDM switching chip sends the second voice digital signal to the POTS; and the POTS converts the received second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
In the existing method for realizing the call, because a certain line concentration ratio exists between the number of users which can be processed by the VOIP chip and the number of users who are concurrently called, when the number of the users who are concurrently called is large and VOIP resources are insufficient, the probability of user call failure is large, and thus the user experience degree is low.
Disclosure of Invention
In order to solve the above problems, the present invention provides a method and an apparatus for implementing a call, which can reduce the probability of a call failure of a user, thereby improving the user experience.
In order to achieve the above object, the present invention provides a method for implementing a call, including:
the uplink card receives a resource allocation request from the soft switch equipment and sends the resource allocation request to the main control board;
the main control board judges that the real-time transmission protocol RTP coding and decoding resources which can be used exist in the POTS of the voice user board corresponding to the user, distributes first RTP coding and decoding resources for the user according to the RTP coding and decoding resources which can be used on the POTS corresponding to the user, and sends the distributed first RTP coding and decoding resources to the POTS corresponding to the user.
Preferably, when the POTS corresponding to the subscriber receives a first voice analog signal from the subscriber, the method further includes:
the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal, the first voice digital signal is coded by adopting the allocated first RTP coding and decoding resources, an Internet Protocol (IP) address and a port are packaged on the coded first voice digital signal to obtain a first IP voice data packet, and the first IP voice data packet is sent to the main control board;
and the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
Preferably, when the upstream card receives the second voice over IP data packet, the method further includes:
the uplink card sends the second IP voice data packet to the main control board;
the main control board sends the second IP voice data packet to a POTS corresponding to the user;
and the POTS corresponding to the user decapsulates the second IP voice data packet, decodes the decapsulated second IP voice data packet by adopting the allocated first RTP encoding and decoding resources to obtain a second voice digital signal, converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
Preferably, when the main control board determines that the POTS corresponding to the user does not have available RTP codec resources, and the main control board has available RTP codec resources, the method further includes:
and the main control board allocates a second RTP coding and decoding resource for the user according to the RTP coding and decoding resource which can be used on the main control board.
Preferably, when the POTS corresponding to the subscriber receives a first voice analog signal from the subscriber, the method further includes:
the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal and sends the first voice digital signal to the main control board;
and the main control board encodes the first voice digital signal by adopting the distributed second RTP coding and decoding resources, encapsulates an Internet Protocol (IP) address and a port on the encoded first voice digital signal to obtain a first IP voice data packet, and sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
Preferably, when the upstream card receives the second voice over IP data packet, the method further includes:
the uplink card sends the second IP voice data packet to the main control board;
the main control board decapsulates the second IP voice data packet, and decodes the decapsulated second IP voice data packet by using the allocated second RTP encoding and decoding resource to obtain a second voice digital signal;
and the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
Preferably, when the main control board determines that neither the POTS corresponding to the user nor the main control board has a usable RTP codec resource, and that there are usable RTP codec resources on other POTS, the method further includes:
and allocating a third RTP coding and decoding resource for the user according to the RTP coding and decoding resources available on the other POTS, and sending the allocated third RTP coding and decoding resource to the POTS corresponding to the allocated third RTP coding and decoding resource.
Preferably, when the POTS corresponding to the subscriber receives a first voice analog signal from the subscriber, the method further includes:
the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal and sends the first voice digital signal to the main control board;
the main control board sends the first voice digital signal to a POTS corresponding to the distributed third RTP coding and decoding resource;
the POTS corresponding to the distributed third RTP coding and decoding resources adopts the distributed third RTP coding and decoding resources to code the first voice digital signal, an Internet Protocol (IP) address and a port are packaged on the coded first voice digital signal to obtain a first IP voice data packet, and the first IP voice data packet is sent to the main control board;
and the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
Preferably, when the upstream card receives the second voice over IP data packet, the method further includes:
the uplink card sends the second IP voice data packet to the main control board;
the main control board sends the second IP voice data packet to the POTS corresponding to the distributed third RTP coding and decoding resource;
the POTS corresponding to the distributed third RTP coding and decoding resources is used for decapsulating the second IP voice data packet, the decapsulated second IP voice data packet is decoded by adopting the distributed third RTP coding and decoding resources to obtain a second voice digital signal, and the second voice digital signal is sent to the main control board;
the main control board sends the second voice digital signal to a POTS corresponding to the user;
and the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
The invention also provides a device for realizing the call, which at least comprises:
the uplink card is used for receiving a resource allocation request from the soft switch equipment and sending the resource allocation request to the main control board;
the main control board is used for judging that the real-time transport protocol (RTP) coding and decoding resources which can be used by the POTS of the voice user board corresponding to the user exist, distributing first RTP coding and decoding resources for the user according to the RTP coding and decoding resources which can be used by the POTS corresponding to the user, and sending the distributed first RTP coding and decoding resources to the POTS corresponding to the user.
Preferably, the method further comprises the following steps:
the POTS corresponding to the user is configured to receive a first voice analog signal from the user, convert the first voice analog signal into a first voice digital signal, encode the first voice digital signal by using the allocated first RTP encoding and decoding resource, encapsulate an internet protocol IP address and a port on the encoded first voice digital signal to obtain a first IP voice data packet, and send the first IP voice data packet to the main control board;
the main control board is also used for:
and sending the IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
Preferably, the add-on card is further configured to:
receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
sending the second IP voice data packet to a POTS (plain old telephone service) corresponding to the user;
the POTS corresponding to the subscriber is further configured to:
and decapsulating the second IP voice data packet, decoding the decapsulated second IP voice data packet by using the allocated first RTP codec resource to obtain a second voice digital signal, converting the second voice digital signal into a second voice analog signal, and sending the second voice analog signal to a user.
Preferably, the main control board is further configured to:
and judging that the POTS corresponding to the user does not have usable RTP coding and decoding resources, wherein the usable RTP coding and decoding resources exist in the main control board, and allocating second RTP coding and decoding resources for the user according to the usable RTP coding and decoding resources on the main control board.
Preferably, the method further comprises the following steps:
the POTS corresponding to the user is used for receiving a first voice analog signal from the user, converting the first voice analog signal into a first voice digital signal and sending the first voice digital signal to the main control board;
the main control board is also used for:
and encoding the first voice digital signal by adopting the distributed second RTP encoding and decoding resources, encapsulating an Internet Protocol (IP) address and a port on the encoded first voice digital signal to obtain a first IP voice data packet, and sending the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
Preferably, the add-on card is further configured to:
receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
decapsulating the second IP voice data packet, and decoding the decapsulated second IP voice data packet by using the allocated second RTP encoding and decoding resource to obtain a second voice digital signal;
the POTS corresponding to the subscriber is further configured to:
and converting the second voice digital signal into a second voice analog signal and then sending the second voice analog signal to a user.
Preferably, the main control board is further configured to:
and judging that neither the POTS corresponding to the user nor the main control board can use the RTP coding and decoding resources, and other POTS has usable RTP coding and decoding resources, allocating third RTP coding and decoding resources for the user according to the usable RTP coding and decoding resources on other POTS, and sending the allocated third RTP coding and decoding resources to the POTS corresponding to the allocated third RTP coding and decoding resources.
Preferably, the method further comprises the following steps:
the POTS corresponding to the user is used for receiving a first voice analog signal from the user, converting the first voice analog signal into a first voice digital signal and sending the first voice digital signal to the main control board;
the POTS corresponding to the allocated third RTP codec resource is configured to encode the first voice digital signal by using the allocated third RTP codec resource, encapsulate an internet protocol IP address and a port for the encoded first voice digital signal to obtain an IP voice data packet, and send the first IP voice data packet to the main control board;
the main control board is also used for:
sending the first voice digital signal to a POTS corresponding to the distributed third RTP coding and decoding resource; and sending the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
Preferably, the add-on card is further configured to:
receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
sending the second IP voice data packet to the POTS corresponding to the distributed third RTP coding and decoding resource; sending the second voice digital signal to a POTS corresponding to the user;
POTS corresponding to the allocated third RTP codec resource is further configured to:
decapsulating the second IP voice data packet, decoding the decapsulated second IP voice data packet by using the allocated third RTP codec resource to obtain a second voice digital signal, and sending the second voice digital signal to the main control board;
the POTS corresponding to the subscriber is further configured to:
and converting the second voice digital signal into a second voice analog signal and then sending the second voice analog signal to a user.
Compared with the prior art, the invention comprises the following steps: the uplink card receives a resource allocation request from the soft switch equipment and sends the resource allocation request to the main control board; the main control board judges that the POTS corresponding to the user has available RTP coding and decoding resources, allocates first RTP coding and decoding resources for the user according to the available RTP coding and decoding resources on the POTS corresponding to the user, and sends the allocated first RTP coding and decoding resources to the POTS corresponding to the user. By the scheme of the invention, when the POTS corresponding to the user has available RTP coding and decoding resources, the RTP coding and decoding resources available on the POTS corresponding to the user are allocated to the user, so that the probability of call failure of the user is reduced, and the experience of the user is improved.
Drawings
The accompanying drawings in the embodiments of the present invention are described below, and the drawings in the embodiments are provided for further understanding of the present invention, and together with the description serve to explain the present invention without limiting the scope of the present invention.
Fig. 1 is a schematic structural component diagram of a conventional device for implementing a call;
FIG. 2 is a flow chart of a method of implementing a call in accordance with the present invention;
FIG. 3 is a schematic structural diagram of a device for implementing a call according to the present invention;
FIG. 4 is a schematic structural diagram of an apparatus for implementing a call according to a first embodiment of the present invention;
FIG. 5 is a schematic structural diagram of an apparatus for implementing a call according to a second embodiment of the present invention;
fig. 6 is a schematic structural diagram of a device for implementing a call according to a third embodiment of the present invention.
Detailed Description
The following further description of the present invention, in order to facilitate understanding of those skilled in the art, is provided in conjunction with the accompanying drawings and is not intended to limit the scope of the present invention. In the present application, the embodiments and various aspects of the embodiments may be combined with each other without conflict.
Referring to fig. 2, the present invention provides a method for implementing a call, including:
step 200, the uplink card receives the resource allocation request from the soft switch device and sends the resource allocation request to the main control board.
Step 201, the main control board determines that there is an available RTP codec resource of the POTS corresponding to the user, allocates a first RTP codec resource to the user according to the available RTP codec resource on the POTS corresponding to the user, and sends the allocated first RTP codec resource to the POTS corresponding to the user.
In this step, the main control board pre-stores the usage of the RTP encoding and decoding resources on each POTS, the usage of the RTP encoding and decoding resources of the main control board, and the corresponding relationship between the user and the POTS.
The main control board can judge whether the POTS corresponding to the user has available RTP coding and decoding resources according to the service condition of the RTP coding and decoding resources on each POTS and the corresponding relationship between the user and the POTS.
In this step, the main control board marks the allocated first RTP codec resource as used.
In this step, after receiving the allocated first RTP codec resource, the POTS corresponding to the user may store a corresponding relationship between the allocated first RTP codec resource and the user.
When the POTS corresponding to the subscriber receives the first voice analog signal from the subscriber, the method further includes:
the POTS corresponding to a user converts a first voice analog signal into a first voice digital signal, the first voice digital signal is coded by adopting a distributed first RTP coding and decoding resource, an IP address and a port are packaged on the coded first voice digital signal to obtain a first IP voice data packet, and the first IP voice data packet is sent to a main control board; and the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
When the uplink card receives the second IP voice data packet, the method further comprises the following steps:
the uplink card sends the second IP voice data packet to the main control board; the main control board sends the second IP voice data packet to a POTS corresponding to the user; and the POTS corresponding to the user decapsulates the second IP voice data packet, decodes the decapsulated second IP voice data packet by adopting the allocated first RTP coding and decoding resources to obtain a second voice digital signal, converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
When the main control board judges that the POTS corresponding to the user does not have usable RTP coding and decoding resources and the main control board has usable RTP coding and decoding resources, the method also comprises the following steps:
and the main control board allocates a second RTP coding and decoding resource for the user according to the RTP coding and decoding resource which can be used on the main control board.
When the POTS corresponding to the subscriber receives a first voice analog signal from the subscriber, the method further includes:
the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal and sends the first voice digital signal to the main control board; the main control board adopts the distributed second RTP coding and decoding resources to code the first voice digital signal, packages an Internet protocol IP address and a port on the coded first voice digital signal to obtain a first IP voice data packet, and sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
When the uplink card receives the second IP voice data packet, the method further comprises the following steps:
the uplink card sends the second IP voice data packet to the main control board; the main control board decapsulates the second IP voice data packet, and decodes the decapsulated second IP voice data packet by adopting a distributed second RTP encoding and decoding resource to obtain a second voice digital signal; and the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
When the main control board judges that neither the POTS corresponding to the user nor the main control board has a usable RTP codec resource, and that usable RTP codec resources exist on other POTS, the method further includes:
and allocating a third RTP coding and decoding resource for the user according to the RTP coding and decoding resources available on other POTS, and sending the allocated third RTP coding and decoding resource to the POTS corresponding to the allocated third RTP coding and decoding resource.
When the POTS corresponding to the subscriber receives a first voice analog signal from the subscriber, the method further includes:
the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal and sends the first voice digital signal to the main control board; the main control board sends the first voice digital signal to a POTS corresponding to the distributed third RTP coding and decoding resource; POTS corresponding to the distributed third RTP coding and decoding resources is used for coding the first voice digital signal by adopting the distributed third RTP coding and decoding resources, an Internet Protocol (IP) address and a port are packaged on the coded first voice digital signal to obtain a first IP voice data packet, and the first IP voice data packet is sent to the main control board; and the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
When the uplink card receives the second IP voice data packet, the method further comprises the following steps:
the uplink card sends the second IP voice data packet to the main control board; the main control board sends the second IP voice data packet to a POTS corresponding to the distributed third RTP coding and decoding resource; the POTS corresponding to the distributed third RTP coding and decoding resources is used for decapsulating the second IP voice data packet, the decapsulated second IP voice data packet is decoded by adopting the distributed third RTP coding and decoding resources to obtain a second voice digital signal, and the second voice digital signal is sent to the main control board; the main control board sends the second voice digital signal to a POTS corresponding to the user; and the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
In the method of the invention, different POTS can be configured for users according to different user grades, POTS containing RTP coding and decoding resources can be configured for high-grade users, and POTS not containing RTP coding and decoding resources can be configured for common users.
Referring to fig. 3, the present invention further provides an apparatus for implementing a call, which at least includes:
the uplink card is used for receiving a resource allocation request from the soft switch equipment and sending the resource allocation request to the main control board;
the main control board is used for judging that the real-time transport protocol (RTP) coding and decoding resources which can be used by the POTS of the voice user board corresponding to the user exist, distributing first RTP coding and decoding resources for the user according to the RTP coding and decoding resources which can be used by the POTS corresponding to the user, and sending the distributed first RTP coding and decoding resources to the POTS corresponding to the user.
The apparatus of the present invention further comprises:
the POTS corresponding to the user is used for receiving a first voice analog signal from the user, converting the first voice analog signal into a first voice digital signal, encoding the first voice digital signal by adopting a distributed first RTP (real time protocol) encoding and decoding resource, packaging an Internet Protocol (IP) address and a port of the encoded first voice digital signal to obtain a first IP voice data packet, and sending the first IP voice data packet to the main control board;
the main control board is also used for:
and sending the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
In the device of the present invention, the add-on card is further configured to:
receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
sending the second IP voice data packet to a POTS (plain old telephone service) corresponding to the user;
the POTS to which the subscriber corresponds is also used to:
and decapsulating the second IP voice data packet, decoding the decapsulated second IP voice data packet by using the allocated first RTP codec resource to obtain a second voice digital signal, converting the second voice digital signal into a second voice analog signal, and sending the second voice analog signal to the user.
In the apparatus of the present invention, the main control board is further configured to:
and judging that the POTS corresponding to the user does not have usable RTP coding and decoding resources, and the main control board has usable RTP coding and decoding resources, and allocating second RTP coding and decoding resources for the user according to the usable RTP coding and decoding resources on the main control board.
The apparatus of the present invention further comprises:
the POTS corresponding to the user is used for receiving a first voice analog signal from the user, converting the first voice analog signal into a first voice digital signal and sending the first voice digital signal to the main control board;
the main control board is also used for:
and encoding the first voice digital signal by adopting the allocated second RTP encoding and decoding resources, encapsulating an Internet Protocol (IP) address and a port on the encoded first voice digital signal to obtain a first IP voice data packet, and sending the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
In the device of the present invention, the add-on card is further configured to:
receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
decapsulating the second IP voice data packet, and decoding the decapsulated second IP voice data packet by using a distributed second RTP encoding and decoding resource to obtain a second voice digital signal;
the POTS to which the subscriber corresponds is also used to:
and converting the second voice digital signal into a second voice analog signal and then sending the second voice analog signal to a user.
In the apparatus of the present invention, the main control board is further configured to:
and judging that neither the POTS corresponding to the user nor the main control board can use the RTP coding and decoding resources, and other POTS has usable RTP coding and decoding resources, allocating third RTP coding and decoding resources for the user according to the usable RTP coding and decoding resources on other POTS, and sending the allocated third RTP coding and decoding resources to the POTS corresponding to the allocated third RTP coding and decoding resources.
The apparatus of the present invention further comprises:
the POTS corresponding to the user is used for receiving a first voice analog signal from the user, converting the first voice analog signal into a first voice digital signal and sending the first voice digital signal to the main control board;
the POTS corresponding to the distributed third RTP coding and decoding resources is used for coding the first voice digital signal by adopting the distributed third RTP coding and decoding resources, obtaining a first IP voice data packet after packaging an Internet protocol IP address and a port of the coded first voice digital signal, and sending the first IP voice data packet to the main control board;
the main control board is also used for:
sending the first voice digital signal to a POTS (plain old telephone service) corresponding to the distributed third RTP coding and decoding resource; and sending the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
In the device of the present invention, the add-on card is further configured to:
receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
sending the second IP voice data packet to a POTS corresponding to the distributed third RTP coding and decoding resource; sending the second voice digital signal to a POTS corresponding to the user;
POTS corresponding to the allocated third RTP codec resource is further configured to:
decapsulating the second IP voice data packet, decoding the decapsulated second IP voice data packet by using a distributed third RTP codec resource to obtain a second voice digital signal, and sending the second voice digital signal to the main control board;
the POTS to which the subscriber corresponds is also used to:
and converting the second voice digital signal into a second voice analog signal and then sending the second voice analog signal to a user.
The method and apparatus of the present invention are described in detail below by way of specific examples.
Fig. 4 is a schematic structural diagram of a device for implementing a call according to a first embodiment. As shown in fig. 4, the method for implementing a call includes:
the uplink card receives a resource allocation request from the soft switch equipment and sends the resource allocation request to an IP switch chip of the main control board; the IP exchange chip of the main control board sends the resource allocation request to the VOIP chip of the main control board;
the VOIP chip of the main control board judges that the POTS of the voice user board corresponding to the user has available real-time transmission protocol RTP coding and decoding resources, distributes first RTP coding and decoding resources for the user according to the available RTP coding and decoding resources on the POTS corresponding to the user, sends the distributed first RTP coding and decoding resources to the TDM switching chip of the main control board, and the TDM switching chip of the main control board sends the distributed first RTP coding and decoding resources to the VOIP chip of the POTS corresponding to the user.
When an Analog To Digital Converter (ADC) chip of the POTS corresponding To the user receives a first voice Analog signal from the user, converting the first voice Analog signal into a first voice Digital signal by the ADC chip of the POTS corresponding To the user, and sending the first voice Digital signal To a VOIP chip of the POTS corresponding To the user; a VOIP chip of POTS corresponding to a user adopts distributed first RTP coding and decoding resources to code a first voice digital signal, an internet protocol IP address and a port are packaged on the coded first voice digital signal to obtain a first IP voice data packet, and the first IP voice data packet is sent to an IP exchange chip of a main control board; and the IP exchange chip of the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
When the uplink card receives the second IP voice data packet, the uplink card sends the second IP voice data packet to the IP exchange chip of the main control board;
the IP exchange chip of the main control board sends the second IP voice data packet to a VOIP chip of POTS corresponding to the user;
the VOIP chip of the POTS corresponding to the user unpacks the second IP voice data packet, the unpacked second IP voice data packet is decoded by adopting the distributed first RTP coding and decoding resources to obtain a second voice digital signal, and the second voice digital signal is sent to the ADC chip of the POTS corresponding to the user; and the ADC chip of the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
Fig. 5 is a schematic structural diagram of a device for implementing a call according to a second embodiment. As shown in fig. 5, the method for implementing a call includes:
the uplink card receives a resource allocation request from the soft switch equipment and sends the resource allocation request to an IP switch chip of the main control board; the IP exchange chip of the main control board sends the resource allocation request to the VOIP chip of the main control board; and the VOIP chip of the main control board judges that the POTS corresponding to the user does not have available RTP coding and decoding resources, the main control board has available RTP coding and decoding resources, and second RTP coding and decoding resources are distributed to the user according to the available RTP coding and decoding resources on the main control board.
When the POTS corresponding to the user receives a first voice analog signal from the user, the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal and sends the first voice digital signal to the TDM switching chip of the main control board; the TDM switching chip of the main control board sends the first voice digital signal to the VOIP chip of the main control board; the VOIP chip of the main control board adopts the distributed second RTP coding and decoding resources to code the first voice digital signal, packages the IP address and the port of the Internet protocol for the coded first voice digital signal to obtain a first IP voice data packet, and sends the first IP voice data packet to the IP exchange chip of the main control board; and the IP exchange chip of the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
When the uplink card receives the second IP voice data packet, the uplink card sends the second IP voice data packet to the IP exchange chip of the main control board; the IP exchange chip of the main control board sends the second IP voice data packet to the VOIP chip of the main control board; the VOIP chip of the main control board decapsulates the second IP voice data packet, decodes the decapsulated second IP voice data packet by adopting the allocated second RTP encoding and decoding resource to obtain a second voice digital signal, and sends the second digital signal to the TDM switching chip of the main control board; the TDM switching chip of the main control board sends the second digital signal to a POTS corresponding to the user; and the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
Third embodiment, fig. 6 is a schematic structural component diagram of a device for implementing a call according to the third embodiment. As shown in fig. 6, the method for implementing a call includes:
the uplink card receives a resource allocation request from the soft switch equipment and sends the resource allocation request to an IP switch chip of the main control board; the IP exchange chip of the main control board sends the resource allocation request to the VOIP chip of the main control board; the VOIP chip of the main control board judges that both POTS corresponding to the user and the main control board have no usable RTP coding and decoding resources, and other POTS has usable RTP coding and decoding resources, allocates a third RTP coding and decoding resource for the user according to the usable RTP coding and decoding resources on other POTS, and sends the allocated third RTP coding and decoding resource to the TDM switching chip of the main control board; and the TDM switching chip of the main control board sends the distributed third RTP coding and decoding resources to POTS corresponding to the distributed third RTP coding and decoding resources.
When the POTS corresponding to the user receives a first voice analog signal from the user, the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal and sends the first voice digital signal to the TDM chip of the main control board; the TDM chip of the main control board sends the first voice digital signal to a POTS corresponding to the distributed third RTP coding and decoding resource; POTS corresponding to the distributed third RTP coding and decoding resources is used for coding the first voice digital signal by adopting the distributed third RTP coding and decoding resources, an internet protocol IP address and a port are packaged on the coded first voice digital signal to obtain a first IP voice data packet, and the first IP voice data packet is sent to an IP exchange chip of the main control board; and the IP exchange chip of the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
When the uplink card receives the second IP voice data packet, the uplink card sends the second IP voice data packet to the IP exchange chip of the main control board; the IP exchange chip of the main control board sends the second IP voice data packet to the POTS corresponding to the distributed third RTP coding and decoding resource; the POTS corresponding to the distributed third RTP coding and decoding resources is used for decapsulating the second IP voice data packet, the decapsulated second IP voice data packet is decoded by adopting the distributed third RTP coding and decoding resources to obtain a second voice digital signal, and the second voice digital signal is sent to the TDM switching chip of the main control board; the TDM switching chip of the main control board sends the second voice digital signal to a POTS corresponding to the user; and the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
It should be noted that the above-mentioned embodiments are only for facilitating the understanding of those skilled in the art, and are not intended to limit the scope of the present invention, and any obvious substitutions, modifications, etc. made by those skilled in the art without departing from the inventive concept of the present invention are within the scope of the present invention.

Claims (18)

1. A method for implementing a call, comprising:
the uplink card receives a resource allocation request from the soft switch equipment and sends the resource allocation request to the main control board;
the main control board judges that usable RTP coding and decoding resources exist in the POTS corresponding to the user according to the service condition of real-time transmission protocol RTP coding and decoding resources on the POTS of each voice user board and the corresponding relation between the user and the POTS, allocates first RTP coding and decoding resources for the user according to the usable RTP coding and decoding resources on the POTS corresponding to the user, and sends the allocated first RTP coding and decoding resources to the POTS corresponding to the user.
2. The method of claim 1, wherein when the POTS to which the subscriber corresponds receives the first voice analog signal from the subscriber, the method further comprises:
the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal, the first voice digital signal is coded by adopting the allocated first RTP coding and decoding resources, an Internet Protocol (IP) address and a port are packaged on the coded first voice digital signal to obtain a first IP voice data packet, and the first IP voice data packet is sent to the main control board;
and the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
3. The method of claim 2, wherein when the upstream card receives the second voice over IP data packet, the method further comprises:
the uplink card sends the second IP voice data packet to the main control board;
the main control board sends the second IP voice data packet to a POTS corresponding to the user;
and the POTS corresponding to the user decapsulates the second IP voice data packet, decodes the decapsulated second IP voice data packet by adopting the allocated first RTP encoding and decoding resources to obtain a second voice digital signal, converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
4. The method of claim 1, wherein when the main control board determines that the POTS corresponding to the user has no available RTP codec resource and the main control board has available RTP codec resource, the method further comprises:
and the main control board allocates a second RTP coding and decoding resource for the user according to the RTP coding and decoding resource which can be used on the main control board.
5. The method of claim 4, wherein when the POTS to which the subscriber corresponds receives the first voice analog signal from the subscriber, the method further comprises:
the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal and sends the first voice digital signal to the main control board;
and the main control board encodes the first voice digital signal by adopting the distributed second RTP coding and decoding resources, encapsulates an Internet Protocol (IP) address and a port on the encoded first voice digital signal to obtain a first IP voice data packet, and sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
6. The method of claim 5, wherein when the upstream card receives the second voice over IP data packet, the method further comprises:
the uplink card sends the second IP voice data packet to the main control board;
the main control board decapsulates the second IP voice data packet, and decodes the decapsulated second IP voice data packet by using the allocated second RTP encoding and decoding resource to obtain a second voice digital signal;
and the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
7. The method of claim 1, wherein when the main control board determines that neither the POTS corresponding to the user nor the main control board has available RTP codec resources, and that there are available RTP codec resources on other POTS, the method further comprises:
and allocating a third RTP coding and decoding resource for the user according to the RTP coding and decoding resources available on the other POTS, and sending the allocated third RTP coding and decoding resource to the POTS corresponding to the allocated third RTP coding and decoding resource.
8. The method of claim 7, wherein when the POTS to which the subscriber corresponds receives the first voice analog signal from the subscriber, the method further comprises:
the POTS corresponding to the user converts the first voice analog signal into a first voice digital signal and sends the first voice digital signal to the main control board;
the main control board sends the first voice digital signal to a POTS corresponding to the distributed third RTP coding and decoding resource;
the POTS corresponding to the distributed third RTP coding and decoding resources adopts the distributed third RTP coding and decoding resources to code the first voice digital signal, an Internet Protocol (IP) address and a port are packaged on the coded first voice digital signal to obtain a first IP voice data packet, and the first IP voice data packet is sent to the main control board;
and the main control board sends the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
9. The method of claim 8, wherein when the upstream card receives the second voice over IP data packet, the method further comprises:
the uplink card sends the second IP voice data packet to the main control board;
the main control board sends the second IP voice data packet to the POTS corresponding to the distributed third RTP coding and decoding resource;
the POTS corresponding to the distributed third RTP coding and decoding resources is used for decapsulating the second IP voice data packet, the decapsulated second IP voice data packet is decoded by adopting the distributed third RTP coding and decoding resources to obtain a second voice digital signal, and the second voice digital signal is sent to the main control board;
the main control board sends the second voice digital signal to a POTS corresponding to the user;
and the POTS corresponding to the user converts the second voice digital signal into a second voice analog signal and then sends the second voice analog signal to the user.
10. An apparatus for enabling a call, comprising at least:
the uplink card is used for receiving a resource allocation request from the soft switch equipment and sending the resource allocation request to the main control board;
and the main control board is used for judging that usable RTP coding and decoding resources exist in the POTS corresponding to the user according to the service condition of real-time transmission protocol RTP coding and decoding resources on the POTS of each voice user board and the corresponding relation between the user and the POTS, allocating first RTP coding and decoding resources for the user according to the usable RTP coding and decoding resources on the POTS corresponding to the user, and sending the allocated first RTP coding and decoding resources to the POTS corresponding to the user.
11. The apparatus of claim 10, further comprising:
the POTS corresponding to the user is configured to receive a first voice analog signal from the user, convert the first voice analog signal into a first voice digital signal, encode the first voice digital signal by using the allocated first RTP encoding and decoding resource, encapsulate an internet protocol IP address and a port on the encoded first voice digital signal to obtain a first IP voice data packet, and send the first IP voice data packet to the main control board;
the main control board is also used for:
and sending the IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
12. The apparatus of claim 11, wherein the add-on card is further configured to: receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
sending the second IP voice data packet to a POTS (plain old telephone service) corresponding to the user;
the POTS corresponding to the subscriber is further configured to:
and decapsulating the second IP voice data packet, decoding the decapsulated second IP voice data packet by using the allocated first RTP codec resource to obtain a second voice digital signal, converting the second voice digital signal into a second voice analog signal, and sending the second voice analog signal to a user.
13. The apparatus of claim 10, wherein the master control board is further configured to:
and judging that the POTS corresponding to the user does not have usable RTP coding and decoding resources, wherein the usable RTP coding and decoding resources exist in the main control board, and allocating second RTP coding and decoding resources for the user according to the usable RTP coding and decoding resources on the main control board.
14. The apparatus of claim 13, further comprising:
the POTS corresponding to the user is used for receiving a first voice analog signal from the user, converting the first voice analog signal into a first voice digital signal and sending the first voice digital signal to the main control board;
the main control board is also used for:
and encoding the first voice digital signal by adopting the distributed second RTP encoding and decoding resources, encapsulating an Internet Protocol (IP) address and a port on the encoded first voice digital signal to obtain a first IP voice data packet, and sending the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
15. The apparatus of claim 14, wherein the add-on card is further configured to:
receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
decapsulating the second IP voice data packet, and decoding the decapsulated second IP voice data packet by using the allocated second RTP encoding and decoding resource to obtain a second voice digital signal;
the POTS corresponding to the subscriber is further configured to:
and converting the second voice digital signal into a second voice analog signal and then sending the second voice analog signal to a user.
16. The apparatus of claim 10, wherein the master control board is further configured to:
and judging that neither the POTS corresponding to the user nor the main control board can use the RTP coding and decoding resources, and other POTS has usable RTP coding and decoding resources, allocating third RTP coding and decoding resources for the user according to the usable RTP coding and decoding resources on other POTS, and sending the allocated third RTP coding and decoding resources to the POTS corresponding to the allocated third RTP coding and decoding resources.
17. The apparatus of claim 16, further comprising:
the POTS corresponding to the user is used for receiving a first voice analog signal from the user, converting the first voice analog signal into a first voice digital signal and sending the first voice digital signal to the main control board;
the POTS corresponding to the allocated third RTP codec resource is configured to encode the first voice digital signal by using the allocated third RTP codec resource, encapsulate an internet protocol IP address and a port for the encoded first voice digital signal to obtain an IP voice data packet, and send the first IP voice data packet to the main control board;
the main control board is also used for:
sending the first voice digital signal to a POTS corresponding to the distributed third RTP coding and decoding resource; and sending the first IP voice data packet to the uplink card according to the IP address and the port in the first IP voice data packet.
18. The apparatus of claim 17, wherein the add-on card is further configured to:
receiving a second IP voice data packet, and sending the second IP voice data packet to the main control board;
the main control board is also used for:
sending the second IP voice data packet to the POTS corresponding to the distributed third RTP coding and decoding resource; sending the second voice digital signal to a POTS corresponding to the user;
POTS corresponding to the allocated third RTP codec resource is further configured to:
decapsulating the second IP voice data packet, decoding the decapsulated second IP voice data packet by using the allocated third RTP codec resource to obtain a second voice digital signal, and sending the second voice digital signal to the main control board;
the POTS corresponding to the subscriber is further configured to:
and converting the second voice digital signal into a second voice analog signal and then sending the second voice analog signal to a user.
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PCT/CN2016/082339 WO2016197796A1 (en) 2015-06-10 2016-05-17 Call implementing method and apparatus
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