WO2018006678A1 - 语音通话方法及装置 - Google Patents

语音通话方法及装置 Download PDF

Info

Publication number
WO2018006678A1
WO2018006678A1 PCT/CN2017/087317 CN2017087317W WO2018006678A1 WO 2018006678 A1 WO2018006678 A1 WO 2018006678A1 CN 2017087317 W CN2017087317 W CN 2017087317W WO 2018006678 A1 WO2018006678 A1 WO 2018006678A1
Authority
WO
WIPO (PCT)
Prior art keywords
information
voice
voice call
server
terminal
Prior art date
Application number
PCT/CN2017/087317
Other languages
English (en)
French (fr)
Inventor
卢林
Original Assignee
腾讯科技（深圳）有限公司
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by 腾讯科技（深圳）有限公司 filed Critical 腾讯科技（深圳）有限公司
Publication of WO2018006678A1 publication Critical patent/WO2018006678A1/zh

Links

Images

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L51/00—User-to-user messaging in packet-switching networks, transmitted according to store-and-forward or real-time protocols, e.g. e-mail
    • H04L51/07—User-to-user messaging in packet-switching networks, transmitted according to store-and-forward or real-time protocols, e.g. e-mail characterised by the inclusion of specific contents
    • H04L51/10—Multimedia information
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/0018—Speech coding using phonetic or linguistical decoding of the source; Reconstruction using text-to-speech synthesis
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/173—Transcoding, i.e. converting between two coded representations avoiding cascaded coding-decoding

Definitions

  • the embodiments of the present invention relate to the field of voice calls, and in particular, to a voice call method and apparatus.
  • a related voice call method includes: a voice call client acquires voice coded information of a called terminal; and receives voice call information; if the voice call information is voice call information from the local end, according to the voice coded information of the called terminal The voice call information is converted into voice call information that the called terminal can support, and the converted voice call information is sent to the called terminal; if the voice call information is voice call information from the called terminal, the voice call information is used. Convert the voice call information that the local end can support.
  • the voice call client needs to perform voice transcoding, and when a new voice coding format occurs, in order to ensure that the voice call client can normally transcode, the voice call client needs to be updated, and the voice call client is in the voice call client. During the update, the voice call client will not be able to make a voice call.
  • the embodiment of the present application provides a voice call method and apparatus.
  • the technical solution is as follows:
  • a voice call method comprising:
  • a voice communication device comprising:
  • a receiving module configured to receive a voice call request sent by the calling terminal, where the voice call request carries an identifier of the called terminal;
  • An acquiring module configured to acquire first voice coding information of the calling terminal, and second voice coding information of the called terminal;
  • the receiving module is further configured to receive voice call information sent by the calling terminal or the called terminal;
  • a conversion module configured to convert the voice call information into voice call information supported by another terminal according to the first voice coded information and the second voice coded information
  • a sending module configured to send the converted voice call information of the conversion module to the another terminal.
  • the background server After receiving the voice call request sent by the calling terminal, the background server obtains the first voice coded information of the calling terminal and the second voice coded information of the called terminal, and then receives the received call terminal or the called terminal to send The voice call information is converted into the voice call information supported by the other terminal according to the first voice coded information and the second voice coded information, and the converted voice call information is sent to another terminal;
  • the voice call client when a new coding type occurs, during a voice call client update, the voice call client cannot perform a voice call; and the background server can directly perform transcoding according to the voice coding information at both ends of the call, without The voice call client is updated to eliminate the effect that the voice call client may not be able to make a voice call when a new coding type occurs.
  • FIG. 1 is a schematic diagram of an implementation environment involved in various embodiments of the present application.
  • FIG. 2 is a flowchart of a voice call method provided by an embodiment of the present application.
  • FIG. 3 is a flowchart of a voice call method according to another embodiment of the present application.
  • FIG. 4A is a flowchart of a voice call method according to another embodiment of the present application.
  • 4B is a schematic diagram of a voice call method according to another embodiment of the present application.
  • 4C is another flowchart of a voice call method according to another embodiment of the present application.
  • 4D is a schematic diagram of updating target voice coding information of a target terminal according to another embodiment of the present application.
  • FIG. 5 is a schematic structural diagram of a voice call apparatus according to an embodiment of the present application.
  • FIG. 6 is a schematic structural diagram of a server according to an embodiment of the present application.
  • VoIP Voice over Internet Protocol: Digitizes the information of analog signals and transmits them in real time on the IP network in the form of data packets.
  • RTP Real-time Transport Protocol
  • PSTN Public Switched Telephone Network
  • PSTN Public Switched Telephone Network
  • SIP Session Initiation Protocol
  • MMTF Internet Engineering Task Force
  • MMUSIC Internet Engineering Task Force
  • Calling terminal A terminal that initiates a call request when the two parties talk.
  • Called terminal A terminal that receives a voice call initiated by the calling terminal when the two parties talk.
  • FIG. 1 illustrates the implementation of the voice call method provided by various embodiments of the present application.
  • the implementation environment includes a calling terminal 110, a backend server 120, a telephone carrier 130, and a called terminal 140.
  • the calling terminal 110 can be a terminal with voice calling capability, for example, can be a mobile phone.
  • the calling terminal 110 is installed with a voice call client 111, and the calling terminal 110 can initiate a voice call with the called terminal 140 through the voice call client 111.
  • the voice call client can initiate a voice call with the called terminal 140 through VoIP.
  • the calling terminal 110 can be connected to the background server 120 through a wireless network.
  • the background server 120 is a background server for providing services to the voice call client 111.
  • the backend server 120 can be connected to the telephone carrier 130 via a wired or wireless network.
  • the background server 120 can be a server or a server cluster composed of multiple servers.
  • the background server 120 may include an RTP server, a transcoding server, and a call server.
  • the RTP server is used to communicate with the telephone carrier 130, which is used to transcode the voice call information
  • the call server is used to receive the call of the calling terminal 110 and initiate a call to the called terminal 130 to the telephone carrier 130.
  • the background server 120 may further include other servers, which is not limited in this embodiment.
  • the telephone carrier 130 can be a mobile, China Unicom, telecommunications or other carrier.
  • the called terminal 140 can also be a terminal with voice calling capability, for example, can be a mobile phone.
  • the voice call client may be installed in the called terminal 140, or the voice call client may not be installed. This embodiment is not limited thereto.
  • the called terminal 140 can be a terminal in the PSTN.
  • the background server 120 can be connected to the calling terminal 110 and connected to the telephone carrier 130 corresponding to the called terminal 140, that is, the background server 120 can connect to the calling terminal 110 and The call between the called terminal 140, therefore, in the solution described in the following embodiments, the background server 120 can obtain the voice coding information of the calling terminal 110 and the called terminal 140, and further according to the voice coding information of the two.
  • the voice call information between the calling terminal 110 and the called terminal 140 is converted so that the voice can be normally talked even if the voice coding information used by the two is different.
  • the encoding of the encoding format is performed by the background server 120, even if a new encoding type occurs, only an update to the background server 120 is required, and there is no need to update the calling terminal 110 or the called terminal 140.
  • the voice call client needs to be updated before being converted, and has better flexibility.
  • FIG. 2 is a flowchart of a method for a voice call provided by an embodiment of the present application. This embodiment is illustrated by using the voice call method in the background server 120 shown in FIG. As shown in FIG. 2, the voice call method may include:
  • Step 201 Receive a voice call request sent by the calling terminal, where the voice call request carries the identifier of the called terminal.
  • Step 202 Acquire first voice coding information of the calling terminal, and second voice coding information of the called terminal.
  • Step 203 Receive voice call information sent by the calling terminal or the called terminal.
  • Step 204 Convert the voice call information into voice call information supported by another terminal according to the first voice coded information and the second voice coded information.
  • the other terminal refers to a terminal that is engaged in a voice call with the terminal that sends the voice call information. For example, if the received voice call information is sent by the calling terminal, the other terminal is the called terminal; otherwise, if the received voice call information is sent by the called terminal, the other terminal is the calling terminal. .
  • Step 205 Send the converted voice call information to another terminal.
  • the background server obtains the first voice coded information of the calling terminal and the second voice coded information of the called terminal after receiving the voice call request sent by the calling terminal. And after receiving the voice call information sent by the calling terminal or the called terminal, the voice call information is converted into the voice call information supported by the other terminal according to the first voice coded information and the second voice coded information. Transmitting the converted voice call information to another terminal; solving the problem that when a new coding type occurs in the related art, the transcoding can be realized only after the voice call client is updated, and the flexibility is poor; The transcoding is directly performed according to the voice coding information at both ends of the call, and the voice call client is not required to be updated, thereby improving the flexibility.
  • FIG. 3 is a flowchart of a method for a voice call provided by an embodiment of the present application. This embodiment is illustrated by using the voice call method in the background server 120 shown in FIG. 1. As shown in FIG. 3, the voice call method may include:
  • Step 301 Receive a voice call request sent by the calling terminal, where the voice call request carries the identifier of the called terminal.
  • a voice call client is installed in the calling terminal.
  • the user can initiate a voice call request to the called terminal through the voice call client in the calling terminal.
  • the background server may receive the voice call request accordingly.
  • the voice call request carries the identifier of the called terminal. For example, the mobile phone number carrying the called terminal.
  • the voice call request may further include first voice coding information supported by the calling terminal.
  • the first voice encoding information may include: an encoding type of the encoder, or an encoding type and an encoding parameter used by the encoder.
  • the coding type may be: silk, g711a, g729a, etc.
  • the coding parameters may include at least one of a sampling rate, an encoding complexity, and a transmission interval for transmitting adjacent data packets.
  • the voice coding information may include only the coding type. If the encoder configures the coding parameters, then at this time, the speech coding information includes the coding type and the coding parameters.
  • the coding type may also be other types, and the coding parameters may also include other content, which is not limited in this embodiment. .
  • Step 302 Extract the first voice coding information carried in the voice call request.
  • Step 303 Acquire second voice coding information from an operator corresponding to the called terminal according to the identifier of the called terminal in the voice call request.
  • the background server may extract the identifier of the called terminal carried in the voice call request, determine the operator corresponding to the called terminal according to the identifier of the called terminal, and then obtain the called terminal from the determined operator. Second speech encoded information.
  • the background server may send the information acquisition request to the operator, and receive the second voice coded information returned by the operator, where the information acquisition request is used to request to acquire the second voice coded information of the called terminal.
  • the background server may determine that the called terminal is a mobile user. At this time, the background server may send an information acquisition request to the mobile operator, and receive the second voice coding information returned by the mobile operator.
  • the voice coding information of each user in the same operator may be the same or different, and the voice coding information of each user in different operators may be the same or different, which is not limited in this embodiment.
  • the information acquisition request sent by the background server may include the identifier of the called terminal. After receiving the information acquisition request, the operator determines the second voice coding information of the called terminal according to the identifier of the called terminal, and returns the determined second voice code. Information to the backend server.
  • Step 304 Receive voice call information sent by the calling terminal or the called terminal.
  • the background server may correspondingly send the voice call information to the voice call client in the calling terminal; and if the called terminal sends a voice, After the called terminal sends the voice call information to the operator, the operator can forward the voice call information to the background server, and the background server receives the voice call information accordingly.
  • Step 305 Convert the voice call information into voice call information supported by another terminal according to the first voice coded information and the second voice coded information.
  • the background server may convert the voice call information into voice call information supported by another terminal.
  • the background server converts the voice call information into the voice call information corresponding to the second voice coded information of the called terminal; and if the voice call information is sent by the called terminal, The information is converted by the background server to the voice call information corresponding to the first voice coded information of the calling terminal.
  • the background server does not need to perform the conversion, and the direct forwarding is not necessary.
  • Step 306 Send the converted voice call information to another terminal.
  • the background server can send the converted voice call information to another terminal.
  • the other terminal After the other terminal receives the converted voice call information, the other terminal can successfully parse the voice call information to ensure the normal progress of the call.
  • Step 307 In the process of the voice call, receiving the coded information update request sent by the target terminal, the target terminal is the calling terminal or the called terminal, and the encoded information update request carries the updated voice coding information.
  • the background server can receive the encoded information update request sent by the target terminal.
  • both parties can monitor the voice quality in real time, according to the voice quality and voice coding Corresponding relationship between the information, the speech coding information corresponding to the current sound quality is obtained, and if the obtained speech coding information is different from the currently used speech coding information, the coding information update request is sent to the background server.
  • the speech coding information that needs to be updated may be an encoding parameter.
  • the coding parameter is coding complexity
  • the sound quality is positively correlated with the coding complexity
  • the coding parameter is the packet transmission interval
  • the sound quality is negatively correlated with the packet transmission interval
  • the coding parameter includes the sampling rate
  • the sound quality and the sampling rate are positive. relationship.
  • the sound quality in a certain range may correspond to the same voice coding information, which is not limited in this embodiment.
  • Step 308 Update the voice coding information corresponding to the target terminal according to the updated voice coding information.
  • the background server After receiving the encoding information update request, the background server updates the corresponding voice encoding information. After that, the background server can perform transcoding according to the updated voice coding information, which is not described herein again in this embodiment.
  • step 307 and step 308 are optional steps, which may or may not be performed in actual implementation, and the embodiment is only performed after step 306, and optionally, it may also be in the step. Any step after 302 is performed, and the embodiment is not described herein again.
  • the calling terminal can send a call end command to the background server, and after receiving the call end command, the background server deletes the first voice coded information of the previously received calling terminal and the called party.
  • the second voice coded information of the terminal is a point to be explained.
  • the background server obtains the first voice coded information of the calling terminal and the second voice coded information of the called terminal after receiving the voice call request sent by the calling terminal. And after receiving the voice call information sent by the calling terminal or the called terminal, the voice call information is converted into the voice call information supported by the other terminal according to the first voice coded information and the second voice coded information.
  • Transmitting the converted voice call information to another terminal solving the problem that the voice call client cannot make a voice call during the update of the voice call client when a new coding type occurs in the related art;
  • the transcoding is directly performed according to the voice coding information at both ends of the call, without updating the voice call client, and eliminating the effect that the voice call client may not be able to make a voice call when a new coding type occurs.
  • the target terminal can update the corresponding voice coding information in the transcoding server, so that both parties can successfully parse after receiving the voice call information of the peer end, ensuring that the call can be performed normally.
  • the background server is a server.
  • the background server may also be a server cluster consisting of an RTP server, a transcoding server, and a call server.
  • the voice call method may include:
  • Step 401 The call server receives a voice call request sent by the calling terminal.
  • the call server can receive the voice call request accordingly.
  • the voice call request carries the first voice coded information of the calling terminal and the identifier of the called terminal.
  • the voice call client can send the voice call request by using SIP signaling.
  • the voice call client can access the call server through signaling.
  • the call server receives the voice call request.
  • Step 402 The call server sends the first voice coding information carried in the voice call request to the RTP server.
  • the call server sends the first voice coded information to the RTP server, and may send the address of the RTP server to the calling terminal, so that the subsequent calling terminal may send the voice call information to the RTP server according to the address of the RTP server.
  • step 403 the RTP server receives the first voice coding information.
  • Step 404 The call server acquires second voice coding information from the operator according to the identifier of the called terminal carried in the voice call request.
  • step 303 is similar to step 303 in the foregoing embodiment, and details are not described herein again.
  • step 405 the call server synchronizes the second voice encoding information to the RTP server.
  • Step 406 The RTP server receives the second voice coding information.
  • Step 407 The RTP server sends the first voice encoding information and the second voice encoding information to the transcoding server.
  • the RTP server may send the first voice encoding information and the second voice encoding information to the transcoding server.
  • the RTP server may obtain the second voice encoding information and obtain the first voice encoding information.
  • the RTP server obtains both at the same time, which is not limited in this embodiment.
  • step 408 the transcoding server feeds back the identification information to the RTP server.
  • the transcoding server After receiving the first voice encoding information and the second voice encoding information, the transcoding server uniquely assigns an identifier information to the first voice encoding information and the second voice encoding information, and feeds back the identifier information to the RTP server.
  • the identifier information is used to uniquely identify a correspondence between the first voice encoding information and the second voice encoding information.
  • Step 409 The RTP server receives the identifier information fed back by the transcoding server.
  • Step 410 The RTP server receives the voice call information sent by the calling terminal or the called terminal.
  • the calling terminal or the called terminal can send voice call information, and correspondingly, the RTP server can receive the voice call information.
  • the voice call client in the calling terminal can directly send the voice call information to the RTP server.
  • the called terminal sends the voice call information
  • the called terminal can send the voice call information to the RTP server through the operator.
  • step 411 the RTP server sends the voice call information and the identification information to the transcoding server.
  • the RTP server may send the voice call information and the identification information to the transcoding server.
  • Step 412 The transcoding server converts the voice call information into the voice call information supported by the other terminal according to the identification information.
  • Step 413 The transcoding server sends the converted voice call information to the RTP server.
  • Step 414 The RTP server sends the converted voice call information to another terminal.
  • Step 415 after the call ends, the RTP server sends a call end command to the transcoding server, and the call end instruction includes the identification information.
  • Step 416 The transcoding server deletes the first voice encoding information and the second voice encoding information corresponding to the identifier information.
  • the transcoding server After receiving the call end instruction, the transcoding server extracts the identification information in the call end instruction, deletes the first voice coding information and the second voice coding information corresponding to the identification information, and releases the storage space required for storing the information. .
  • the calling terminal or the called terminal may request to update its own voice coding information.
  • the voice call method may further include the following steps:
  • Step 417 the RTP server receives the encoded information update request sent by the target terminal.
  • the call server may forward the encoded information update request to The RTP server receives the encoded information update request sent by the call server correspondingly.
  • the called terminal may send the encoded information update request to the call server, and the call server forwards the encoded information update request to the RTP server, and correspondingly, the RTP server receives the encoded information update forwarded by the call server. request.
  • step 418 the RTP server forwards the encoded information update request to the transcoding server.
  • Step 419 The transcoding server updates the speech coding information of the target terminal according to the updated speech coding information in the coding information update request.
  • FIG. 4D shows a schematic diagram of a speech encoding information update process.
  • the background server obtains the first voice coded information of the calling terminal and the second voice coded information of the called terminal after receiving the voice call request sent by the calling terminal. And after receiving the voice call information sent by the calling terminal or the called terminal, the voice call information is converted into the voice call information supported by the other terminal according to the first voice coded information and the second voice coded information.
  • Transmitting the converted voice call information to another terminal solving the problem that the voice call client cannot make a voice call during the update of the voice call client when a new coding type occurs in the related art;
  • the transcoding is directly performed according to the voice coding information at both ends of the call, without updating the voice call client, and eliminating the effect that the voice call client may not be able to make a voice call when a new coding type occurs.
  • the transcoding server After receiving the first voice encoding information and the second voice encoding information, the transcoding server allocates an identifier information for indicating the correspondence between the two, and feeds back the identifier information to the RTP server, so that the RTP server receives the voice call information of one end. After that, only the voice call information and the identification information need to be sent to the transcoding server to implement transcoding, without sending the first voice encoding information and the second voice encoding information to the transcoding server every time, thereby reducing the transmission process.
  • the transmission resources that are required to be used.
  • the target terminal can update the corresponding voice coding information in the transcoding server, so that both parties can successfully parse after receiving the voice call information of the peer end, ensuring that the call can be performed normally.
  • FIG. 5 is a schematic structural diagram of a voice communication device according to an embodiment of the present disclosure.
  • the voice communication device may include: a receiving module 510 , an obtaining module 520 , a converting module 530 , and a sending module . 540.
  • the receiving module 510 is configured to receive a voice call request sent by the calling terminal, where the voice call request carries an identifier of the called terminal;
  • An obtaining module 520 configured to acquire first voice coding information of the calling terminal, and second voice coding information of the called terminal;
  • the receiving module 510 is further configured to receive voice call information sent by the calling terminal or the called terminal;
  • the converting module 530 is configured to convert the voice call information into voice call information supported by another terminal according to the first voice coded information and the second voice coded information;
  • the sending module 540 is configured to send the converted voice call information by the converting module 530 to the other terminal.
  • the voice call apparatus obtains the first voice coded information of the calling terminal and the second voice coded information of the called terminal after receiving the voice call request sent by the calling terminal.
  • the voice call information is converted into the voice call information supported by the other terminal according to the first voice coded information and the second voice coded information, and the transmission is converted.
  • the voice call information is sent to another terminal; when the new coding type occurs in the related art, the voice call client cannot make a voice call during the update of the voice call client; and the background server can be based on the call.
  • the voice coding information at both ends is directly transcoded without updating the voice call client, eliminating the effect that the voice call client may not be able to make a voice call when a new coding type occurs.
  • the acquiring module 520 is further configured to extract the first voice coding information carried in the voice call request.
  • the obtaining module 520 is further configured to acquire the second voice coding information from an operator corresponding to the called terminal according to the identifier of the called terminal in the voice call request.
  • the device is used in a background server, where the background server includes: a real-time transport protocol RTP module and a transcoding module;
  • the obtaining module 520 is further configured to:
  • the identifier information is sent to the RTP module by the transcoding module, where the identifier information is used to uniquely identify a correspondence between the first voice encoding information and the second voice encoding information;
  • the receiving module 510 is further configured to receive the voice call information by using the RTP module;
  • the transcoding module 530 is further configured to:
  • the sending module 540 is further configured to send a call end command to the transcoding server by using the RTP server after the end of the call, where the call end instruction includes the identifier information;
  • the device also includes:
  • a deleting module configured to delete, by the transcoding server, the first voice encoding information and the second voice encoding information corresponding to the identifier information.
  • the receiving module 510 is further configured to receive, during a voice call, an encoding information update request sent by the target terminal, where the target terminal is the calling terminal or the called terminal, and the encoding information
  • the updated request carries the updated voice coding information
  • the device also includes:
  • an update module configured to update the voice coding information corresponding to the target terminal according to the updated voice coding information.
  • the RTP module in this embodiment may be formed as an RTP server, and the transcoding module may be formed into a transcoding server, which is not limited in this embodiment.
  • the voice call device provided by the foregoing embodiment is only illustrated by the division of each functional module.
  • the function distribution may be completed by different functional modules according to requirements, that is, the internal structure of the device. Divided into different functional modules to complete all or part of the functions described above.
  • the voice call device and the voice call method provided by the foregoing embodiments are in the same concept, and the specific implementation process is described in detail in the method embodiment, and details are not described herein again.
  • non-transitory computer readable storage medium comprising instructions, such as a memory comprising instructions executable by a processor in a server to perform the voice call method described above.
  • the non-transitory computer readable storage medium may be a ROM, a random access memory (RAM), a CD-ROM, a magnetic tape, a floppy disk, and an optical data storage device.
  • FIG. 6 is a schematic structural diagram of a server provided by an embodiment of the present application.
  • the server is used to implement the voice call method provided in the above embodiment. Specifically:
  • the server 600 includes a central processing unit (CPU) 601, a system memory 604 including a random access memory (RAM) 602 and a read only memory (ROM) 603, and a system bus 605 that connects the system memory 604 and the central processing unit 601.
  • the server 600 also includes a basic input/output system (I/O system) 606 that facilitates transfer of information between various devices within the computer, and mass storage for storing the operating system 613, applications 614, and other program modules 615.
  • I/O system basic input/output system
  • the basic input/output system 606 includes a display 608 for displaying information and an input device 609 such as a mouse or keyboard for user input of information.
  • the display 608 and input device 609 are both connected to the central processing unit 601 via an input and output controller 610 that is coupled to the system bus 605.
  • the basic input/output system 606 can also include an input output controller 610 for receiving and processing input from a plurality of other devices, such as a keyboard, mouse, or electronic stylus.
  • input and output controller 610 also provides output to a display screen, printer, or other type of output device.
  • the mass storage device 607 is connected to the central processing unit 601 by a mass storage controller (not shown) connected to the system bus 605.
  • the mass storage device 607 and its associated computer readable medium provide non-volatile storage for the server 600. That is, the mass storage device 607 can include a computer readable medium (not shown) such as a hard disk or a CD-ROM drive.
  • the computer readable medium can include computer storage media and communication media.
  • Computer storage media includes volatile and nonvolatile, removable and non-removable media implemented in any method or technology for storage of information such as computer readable instructions, data structures, program modules or other data.
  • Computer storage media include RAM, ROM, EPROM, EEPROM, flash memory or other solid state storage technologies, CD-ROM, DVD or other optical storage, tape cartridges, magnetic tape, magnetic disk storage or other magnetic storage devices.
  • RAM random access memory
  • ROM read only memory
  • EPROM Erasable programmable read-only memory
  • EEPROM electrically erasable programmable read-only memory
  • the server 600 may also be operated by a remote computer connected to the network through a network such as the Internet. That is, the server 600 can be connected to the network 612 through a network interface unit 611 connected to the system bus 605, or can also be connected to other types of networks or remote computer systems (not shown) using the network interface unit 611. .
  • the memory also includes one or more programs, the one or more programs being stored in a memory and configured to be executed by one or more processors.
  • the one or more programs described above include instructions for executing the voice call method on the server side described above.
  • a person skilled in the art may understand that all or part of the steps of implementing the above embodiments may be completed by hardware, or may be instructed by a program to execute related hardware, and the program may be stored in a computer readable storage medium.
  • the storage medium mentioned may be a read only memory, a magnetic disk or an optical disk or the like.

Landscapes

  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Computational Linguistics (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Telephonic Communication Services (AREA)

Abstract

一种语音通话方法及装置，该方法包括：接收主叫终端发送的语音呼叫请求，语音呼叫请求中携带有被叫终端的标识（201）；获取主叫终端的第一语音编码信息，以及，被叫终端的第二语音编码信息（202）；接收主叫终端或者被叫终端发送的语音通话信息（203）；根据第一语音编码信息和第二语音编码信息，将语音通话信息转换为另一终端所支持的语音通话信息（204）；发送转换后的语音通话信息至另一终端（205）。解决了相关技术中当出现新的编码类型时，在对语音通话客户端更新期间，语音通话客户端无法进行语音通话的问题。

Description

语音通话方法及装置
本申请要求于2016年7月8日提交中国专利局、申请号为201610539161.7、发明名称为“语音通话方法及装置”的中国专利申请的优先权，其全部内容通过引用结合在本申请中。
技术领域
本申请实施例涉及语音通话技术领域，特别涉及一种语音通话方法及装置。
背景技术
语音通话已经成为人们沟通交流时常用的一种交流方式之一。
相关的一种语音通话方法包括：语音通话客户端获取被叫终端的语音编码信息；接收语音通话信息；若该语音通话信息为来自本端的语音通话信息，则根据被叫终端的语音编码信息将该语音通话信息转换为被叫终端能够支持的语音通话信息，并将转换后的语音通话信息发送至被叫终端；若语音通话信息为来自被叫终端的语音通话信息，则将该语音通话信息转换为本端能够支持的语音通话信息。
发明人在实现本申请实施例的过程中，发现相关技术至少存在以下问题：
上述方法中语音通话客户端需要执行语音转码，而当出现新的语音编码格式时，为了保证语音通话客户端能够正常转码，需要对该语音通话客户端进行更新，而在语音通话客户端更新期间，语音通话客户端将不能进行语音通话。
发明内容
为了解决相关技术中灵活度较差的问题，本申请实施例提供了一种语音通话方法及装置。所述技术方案如下：
第一方面，提供了一种语音通话方法，所述方法包括：
接收主叫终端发送的语音呼叫请求，所述语音呼叫请求中携带有被叫终端的标识；
获取所述主叫终端的第一语音编码信息，以及，所述被叫终端的第二语音 编码信息；
接收所述主叫终端或者所述被叫终端发送的语音通话信息；
根据所述第一语音编码信息和所述第二语音编码信息，将所述语音通话信息转换为另一终端所支持的语音通话信息；
发送转换后的所述语音通话信息至所述另一终端。
第二方面，提供了一种语音通话装置，所述方法包括：
接收模块，用于接收主叫终端发送的语音呼叫请求，所述语音呼叫请求中携带有被叫终端的标识；
获取模块，用于获取所述主叫终端的第一语音编码信息，以及，所述被叫终端的第二语音编码信息；
所述接收模块，还用于接收所述主叫终端或者所述被叫终端发送的语音通话信息；
转换模块，用于根据所述第一语音编码信息和所述第二语音编码信息，将所述语音通话信息转换为另一终端所支持的语音通话信息；
发送模块，用于发送所述转换模块转换后的所述语音通话信息至所述另一终端。
本申请实施例提供的技术方案带来的有益效果包括：
后台服务器通过在接收到主叫终端发送的语音呼叫请求之后，获取主叫终端的第一语音编码信息以及被叫终端的第二语音编码信息，进而在此后接收到主叫终端或者被叫终端发送的语音通话信息时，根据该第一语音编码信息和第二语音编码信息将该语音通话信息转换为另一终端所支持的语音通话信息，发送转换后的语音通话信息至另一终端；解决了相关技术中当出现新的编码类型时，在对语音通话客户端更新期间，语音通话客户端无法进行语音通话的问题；达到了后台服务器可以根据通话两端的语音编码信息直接进行转码，而无需对语音通话客户端进行更新，消除在出现新的编码类型时语音通话客户端可能不能进行语音通话的效果。
附图说明
为了更清楚地说明本申请实施例中的技术方案，下面将对实施例描述中所需要使用的附图作简单地介绍，显而易见地，下面描述中的附图仅仅是本申请的一些实施例，对于本领域普通技术人员来讲，在不付出创造性劳动的前提下， 还可以根据这些附图获得其他的附图。
图1是本申请各个实施例所涉及的实施环境的示意图；
图2是本申请一个实施例提供的语音通话方法的流程图；
图3是本申请另一实施例提供的语音通话方法的流程图；
图4A是本申请另一实施例提供的语音通话方法的流程图；
图4B是本申请另一实施例提供的语音通话方法的示意图；
图4C是本申请另一实施例提供的语音通话方法的另一流程图；
图4D是本申请另一实施例提供的目标终端更新语音编码信息的示意图；
图5是本申请一个实施例提供的语音通话装置的结构示意图；
图6是本申请一个实施例提供的服务器的结构示意图。
具体实施方式
为使本申请的目的、技术方案和优点更加清楚，下面将结合附图对本申请实施方式作进一步地详细描述。
为了便于理解，首先对本申请各个实施例所涉及的术语做简单说明。
VoIP(Voice over Internet Protocol，网络电话)：将模拟信号的信息数字化，以数据封包(Data Packet)的形式在IP网络(IP Network)上做实时传递。
RTP(Real-time Transport Protocol，实时传输协议)：是一个网络传输协议，说明了在互联网上传递音频和视频的标准数据包格式。
PSTN(Public Switched Telephone Network，公共交换电话网络)：是一种用于全球语音通信的电路交换网络。
SIP(Session Initiation Protocol，会话初始协议):是一个由IETF(Internet Engineering Task Force，互联网工程任务小组)MMUSIC工作组开发的协议，作为标准被提议用于创建、修改和终止包括视频、语音、即时通信、在线游戏和虚拟现实等多种多媒体元素在内的交互式用户会话。
IETF:负责互联网标准的开发和推动。
主叫终端:双方通话时主动发起呼叫请求的终端。
被叫终端:双方通话时接听主叫终端发起的语音通话的终端。
请参考图1，其示出了本申请各个实施例提供的语音通话方法所涉及的实 施环境的示意图。如图1所示，该实施环境包括主叫终端110、后台服务器120、电话运营商130和被叫终端140。
主叫终端110可以为具备语音通话能力的终端，比如，可以为手机。实际实现时，主叫终端110中安装有语音通话客户端111，主叫终端110可以通过该语音通话客户端111发起与被叫终端140之间的语音通话。可选的，语音通话客户端可以通过VoIP发起与被叫终端140之间的语音通话。其中，主叫终端110可以通过无线网络与后台服务器120连接。
后台服务器120是用于为语音通话客户端111提供服务的后台服务器。该后台服务器120可以通过有线或者无线网络与电话运营商130连接。实际实现时，该后台服务器120可以为一台服务器，也可以为由多台服务器组成的服务器集群。
以后台服务器120为服务器集群为例，该后台服务器120可以包括RTP服务器、转码服务器和呼叫服务器。RTP服务器用于与电话运营商130通信，转码服务器用于对语音通话信息进行转码，呼叫服务器用于接收主叫终端110的呼叫并向电话运营商130发起对被叫终端130的呼叫。可选的，后台服务器120还可以包括其他服务器，本实施例对此并不做限定。
电话运营商130可以为移动、联通、电信或者其他运营商。
被叫终端140也可以为具备语音通话能力的终端，比如，可以为手机。实际实现时，被叫终端140中可以安装有语音通话客户端，也可以未安装有语音通话客户端，本实施例对此并不做限定。并且，该被叫终端140可以为PSTN中的终端。
在图1所示的实施环境中，由于后台服务器120可以与主叫终端110连接，且与被叫终端140所对应的电话运营商130连接，也即该后台服务器120可以连接主叫终端110和被叫终端140之间的通话，因此，在下述各个实施例所述的方案中，后台服务器120即可获取主叫终端110和被叫终端140的语音编码信息，进而根据两者的语音编码信息对主叫终端110和被叫终端140之间的语音通话信息进行转换，使得即使两者使用的语音编码信息不同两者仍然可以正常通话。另外，由于通过后台服务器120做编码格式的转换，因此即使出现新的编码类型，也仅需要对后台服务器120是适应性的更新即可，而无需对主叫终端110或者被叫终端140做更新，这相比于相关技术中在出现新的编码类型时需要对语音通话客户端更新之后才能转换，具有更好的灵活度。
请参考图2，其示出了本申请一个实施例提供的语音通话方法的方法流程图，本实施例以该语音通话方法用于图1所示的后台服务器120中来举例说明。如图2所示，该语音通话方法可以包括：
步骤201，接收主叫终端发送的语音呼叫请求，语音呼叫请求中携带有被叫终端的标识。
步骤202，获取主叫终端的第一语音编码信息，以及，被叫终端的第二语音编码信息。
步骤203，接收主叫终端或者被叫终端发送的语音通话信息。
步骤204，根据第一语音编码信息和第二语音编码信息，将语音通话信息转换为另一终端所支持的语音通话信息。
其中，另一终端是指与发送语音通话信息的终端正在进行语音通话的终端。比如，接收到的语音通话信息为主叫终端发送，则该另一终端即为被叫终端；反之，若接收到的语音通话信息为被叫终端发送，则该另一终端即为主叫终端。
步骤205，发送转换后的语音通话信息至另一终端。
综上所述，本实施例提供的语音通话方法，后台服务器通过在接收到主叫终端发送的语音呼叫请求之后，获取主叫终端的第一语音编码信息以及被叫终端的第二语音编码信息，进而在此后接收到主叫终端或者被叫终端发送的语音通话信息时，根据该第一语音编码信息和第二语音编码信息将该语音通话信息转换为另一终端所支持的语音通话信息，发送转换后的语音通话信息至另一终端；解决了相关技术中当出现新的编码类型时，只有对语音通话客户端更新之后才能实现转码，灵活度较差的问题；达到了后台服务器可以根据通话两端的语音编码信息直接进行转码，而无需对语音通话客户端进行更新，提高灵活度的效果。
请参考图3，其示出了本申请一个实施例提供的语音通话方法的方法流程图，本实施例以该语音通话方法用于图1所示的后台服务器120中来举例说明。如图3所示，该语音通话方法可以包括：
步骤301，接收主叫终端发送的语音呼叫请求，语音呼叫请求中携带有被叫终端的标识。
主叫终端中安装有语音通话客户端，当用户需要与其他用户进行语音通话时，用户可以通过主叫终端中的该语音通话客户端发起呼叫被叫终端的语音呼叫请求。
在主叫终端中的语音通话客户端发起语音呼叫请求之后，后台服务器可以相应的接收到该语音呼叫请求。其中，语音呼叫请求中携带有被叫终端的标识。比如，携带有被叫终端的手机号。
实际实现时，语音呼叫请求中还可以包括主叫终端所支持的第一语音编码信息。其中，第一语音编码信息可以包括：编码器的编码类型，或者，编码类型和编码器所使用的编码参数。编码类型可以为：silk、g711a、g729a等等，编码参数可以包括采样率、编码复杂度以及发送相邻数据包的发送间隔中的至少一种。
可选的，由于实际实现时，一些编码器并不会配置编码参数，因此，对于此种情况，其语音编码信息可以只包括编码类型。而若编码器配置编码参数，则此时，语音编码信息中包括编码类型和编码参数。
需要说明的是，上述只是以编码类型和编码参数分别为上述内容来举例，可选的，编码类型还可以为其他类型，且编码参数还可以包括其他内容，本实施例对此并不做限定。
步骤302，提取语音呼叫请求中携带的第一语音编码信息。
步骤303，根据语音呼叫请求中的被叫终端的标识，从被叫终端所对应的运营商处获取第二语音编码信息。
后台服务器接收到语音呼叫请求之后，可以提取语音呼叫请求中携带的被叫终端的标识，根据被叫终端的标识确定被叫终端所对应的运营商，然后从确定的运营商处获取被叫终端的第二语音编码信息。可选的，后台服务器可以发送信息获取请求至运营商，接收运营商返回的第二语音编码信息，该信息获取请求用于请求获取被叫终端的第二语音编码信息。
比如，被叫终端的标识为158616xxx12，则后台服务器可以确定被叫终端为移动用户，此时，后台服务器即可发送信息获取请求至移动运营商，接收移动运营商返回的第二语音编码信息。
需要说明的是，同一运营商中的各个用户的语音编码信息可以相同或不同，且不同运营商中的各个用户的语音编码信息也可以相同或不同，本实施例对此并不做限定。并且，当同一运营商中的各个用户的语音编码信息不同时， 后台服务器发送的信息获取请求中可以包括被叫终端的标识，运营商接收到该信息获取请求之后，根据被叫终端的标识确定被叫终端的第二语音编码信息，返回确定的第二语音编码信息至后台服务器。
步骤304，接收主叫终端或者被叫终端发送的语音通话信息。
主叫终端与被叫终端的语音通话过程中，若主叫终端发出语音，则后台服务器可以相应的到主叫终端中的语音通话客户端发送的语音通话信息；而若被叫终端发出语音，则被叫终端将语音通话信息发送至运营商之后，运营商可以转发该语音通话信息至后台服务器，后台服务器相应的接收到该语音通话信息。
步骤305，根据第一语音编码信息和第二语音编码信息，将语音通话信息转换为另一终端所支持的语音通话信息。
在后台服务器接收到该语音通话信息之后，后台服务器可以将该语音通话信息转换为另一终端所支持的语音通话信息。
比如，若语音通话信息为主叫终端发送的信息，则后台服务器将该语音通话信息转换为被叫终端的第二语音编码信息所对应的语音通话信息；而若语音通话信息为被叫终端发送的信息，则后台服务器将该语音通话信息转换为主叫终端的第一语音编码信息所对应的语音通话信息。
需要说明的是，如果第一语音编码信息与第二语音编码信息相同，则后台服务器无需进行转换，直接转发即可，本实施例在此不再赘述。
步骤306，发送转换后的语音通话信息至另一终端。
在转换过后，后台服务器可以发送转换后的语音通话信息至另一终端。
另一终端接收到转换后的语音通话信息之后，另一终端即可成功解析该语音通话信息，保证了通话的正常进行。
步骤307，在语音通话过程中，接收目标终端发送的编码信息更新请求，目标终端为主叫终端或者被叫终端，编码信息更新请求中携带有更新后的语音编码信息。
在语音通话过程中，随着通话网络的变化可能会出现网络时延、网络抖动或者网络丢包等问题，而为了避免该问题，通话双方中的任一方可以自动更新自身的语音编码信息，并发送编码信息更新请求至后台服务器。相应的，后台服务器可以接收到目标终端发送的编码信息更新请求。
在通话过程中，通话双方可以实时监测通话音质，根据音质与语音编码信 息之间的对应关系，获取当前音质所对应的语音编码信息，若获取到的语音编码信息不同于当前使用的语音编码信息，则发送编码信息更新请求至后台服务器。
由于编码类型通常不会变化，因此，实际实现时，需要更新的语音编码信息可以为编码参数。并且，当编码参数为编码复杂度时，音质与编码复杂度呈正相关关系；当编码参数为发包间隔时，音质与发包间隔呈负相关关系；当编码参数包括采样率时，音质与采样率呈正相关关系。可选的，一定范围内的音质可以对应相同的语音编码信息，本实施例对此并不做限定。
步骤308，根据更新后的语音编码信息更新目标终端所对应的语音编码信息。
后台服务器接收到编码信息更新请求之后，更新对应的语音编码信息。此后，后台服务器即可根据更新后的语音编码信息进行转码，本实施例在此不再赘述。
需要说明的一点是，步骤307和步骤308为可选步骤，实际实现时可以执行也可以不执行，并且，本实施例只是以在步骤306之后执行为例，可选的，其还可以在步骤302之后的任一步骤执行，本实施例在此不再赘述。
需要说明的另一点是，在通话结束之后，主叫终端可以发送通话结束指令至后台服务器，后台服务器接收到通话结束指令之后，删除之前接收到的主叫终端的第一语音编码信息以及被叫终端的第二语音编码信息。
综上所述，本实施例提供的语音通话方法，后台服务器通过在接收到主叫终端发送的语音呼叫请求之后，获取主叫终端的第一语音编码信息以及被叫终端的第二语音编码信息，进而在此后接收到主叫终端或者被叫终端发送的语音通话信息时，根据该第一语音编码信息和第二语音编码信息将该语音通话信息转换为另一终端所支持的语音通话信息，发送转换后的语音通话信息至另一终端；解决了相关技术中当出现新的编码类型时，在对语音通话客户端更新期间，语音通话客户端无法进行语音通话的问题；达到了后台服务器可以根据通话两端的语音编码信息直接进行转码，而无需对语音通话客户端进行更新，消除在出现新的编码类型时语音通话客户端可能不能进行语音通话的效果。
同时，目标终端可以更新转码服务器中其所对应的语音编码信息，使得通话双方接收到对端的语音通话信息之后，均能成功解析，保证了通话能够正常进行。
上述实施例只是以该语音通话方法用于后台服务器，且后台服务器为一台服务器来举例。可选的，该后台服务器还可以为由RTP服务器、转码服务器和呼叫服务器组成的服务器集群，此时，请参考图4A，该语音通话方法可以包括：
步骤401，呼叫服务器接收主叫终端发送的语音呼叫请求。
主叫终端中的语音通话客户端发出语音呼叫请求之后，呼叫服务器可以相应的接收到该语音呼叫请求。其中，该语音呼叫请求中携带有主叫终端的第一语音编码信息以及被叫终端的标识。可选的，语音通话客户端可以通过SIP信令发送该语音呼叫请求。
如图4B所示，语音通话客户端可以通过信令接入至呼叫服务器。相应的，呼叫服务器接收到该语音呼叫请求。
步骤402，呼叫服务器发送语音呼叫请求中携带的第一语音编码信息至RTP服务器。
可选的，呼叫服务器发送第一语音编码信息至RTP服务器的同时，可以发送RTP服务器的地址至主叫终端，以便后续主叫终端可以根据RTP服务器的地址发送语音通话信息至该RTP服务器。
步骤403，RTP服务器接收第一语音编码信息。
步骤404，呼叫服务器根据语音呼叫请求中携带的被叫终端的标识从运营商处获取第二语音编码信息。
本步骤与上述实施例中的步骤303类似，本实施例在此不再赘述。
步骤405，呼叫服务器将第二语音编码信息同步至RTP服务器。
步骤406，RTP服务器接收第二语音编码信息。
步骤407，RTP服务器发送第一语音编码信息和第二语音编码信息至转码服务器。
在RTP服务器获取到第一语音编码信息以及第二语音编码信息之后，RTP服务器可以发送该第一语音编码信息以及第二语音编码信息至转码服务器。
需要说明的是，上述只是以RTP服务器先获取第一语音编码信息，然后获取第二语音编码信息为例，可选的，RTP服务器还可以先获取第二语音编码信息后获取第一语音编码信息，或者，RTP服务器同时获取两者，本实施例对此并不做限定。
步骤408，转码服务器反馈标识信息至RTP服务器。
转码服务器接收到第一语音编码信息和第二语音编码信息之后，为该第一语音编码信息和第二语音编码信息唯一分配一个标识信息，并反馈该标识信息至RTP服务器。其中，该标识信息用于唯一标识第一语音编码信息和第二语音编码信息之间的对应关系。
步骤409，RTP服务器接收转码服务器反馈的标识信息。
步骤410，RTP服务器接收主叫终端或者被叫终端发送的语音通话信息。
在通话过程中，主叫终端或者被叫终端可以发送语音通话信息，相应的，RTP服务器可以接收该语音通话信息。
具体的，当主叫终端发送语音通话信息时，主叫终端中的语音通话客户端可以直接发送该语音通话信息至RTP服务器。而当被叫终端发送语音通话信息时，被叫终端可以通过运营商将该语音通话信息发送至RTP服务器。
步骤411，RTP服务器发送语音通话信息以及标识信息至转码服务器。
在RTP服务器接收到语音通话信息之后，RTP服务器可以发送该语音通话信息以及标识信息至转码服务器。
步骤412，转码服务器根据标识信息，将语音通话信息转换为另一终端所支持的语音通话信息。
步骤413，转码服务器发送转换后的语音通话信息至RTP服务器。
步骤414，RTP服务器将转换后的语音通话信息发送至另一终端。
步骤415，在通话结束之后，RTP服务器发送通话结束指令至转码服务器，通话结束指令中包括标识信息。
步骤416，转码服务器删除标识信息所对应的第一语音编码信息和第二语音编码信息。
转码服务器接收到通话结束指令之后，提取该通话结束指令中的标识信息，删除该标识信息所对应的第一语音编码信息以及第二语音编码信息，释放了存储上述信息时所需的存储空间。
另外，与上述实施例类似的是，主叫终端或者被叫终端可以请求更新自己的语音编码信息，此时，请参考图4C，该语音通话方法还可以包括如下步骤：
步骤417，RTP服务器接收目标终端发送的编码信息更新请求。
可选的，当目标终端为主叫终端时，在主叫终端通过信令接入发送该编码信息更新请求至呼叫服务器之后，呼叫服务器可以转发该编码信息更新请求至 RTP服务器，RTP服务器相应的接收呼叫服务器发送的该编码信息更新请求。而当目标终端为被叫终端时，被叫终端可以发送该编码信息更新请求至呼叫服务器，呼叫服务器转发该编码信息更新请求至RTP服务器，相应的，RTP服务器接收呼叫服务器转发的该编码信息更新请求。
步骤418，RTP服务器转发编码信息更新请求至转码服务器。
步骤419，转码服务器根据编码信息更新请求中的更新后的语音编码信息更新目标终端的语音编码信息。
请参考图4D，其示出了语音编码信息更新过程的示意图。
综上所述，本实施例提供的语音通话方法，后台服务器通过在接收到主叫终端发送的语音呼叫请求之后，获取主叫终端的第一语音编码信息以及被叫终端的第二语音编码信息，进而在此后接收到主叫终端或者被叫终端发送的语音通话信息时，根据该第一语音编码信息和第二语音编码信息将该语音通话信息转换为另一终端所支持的语音通话信息，发送转换后的语音通话信息至另一终端；解决了相关技术中当出现新的编码类型时，在对语音通话客户端更新期间，语音通话客户端无法进行语音通话的问题；达到了后台服务器可以根据通话两端的语音编码信息直接进行转码，而无需对语音通话客户端进行更新，消除在出现新的编码类型时语音通话客户端可能不能进行语音通话的效果。
转码服务器在接收到第一语音编码信息和第二语音编码信息之后，分配一个用于表示两者对应关系的标识信息，反馈该标识信息至RTP服务器，使得RTP服务器接收到一端的语音通话信息之后，只需要将语音通话信息和该标识信息发送至转码服务器即可实现转码，而无需每次都发送第一语音编码信息和第二语音编码信息至转码服务器，降低了传输过程中所需耗用的传输资源。
同时，目标终端可以更新转码服务器中其所对应的语音编码信息，使得通话双方接收到对端的语音通话信息之后，均能成功解析，保证了通话能够正常进行。
请参考图5，其示出了本申请一个实施例提供的语音通话装置的结构示意图，如图5所示，该语音通话装置可以包括：接收模块510、获取模块520、转换模块530和发送模块540。
接收模块510，用于接收主叫终端发送的语音呼叫请求，所述语音呼叫请求中携带有被叫终端的标识；
获取模块520，用于获取所述主叫终端的第一语音编码信息，以及，所述被叫终端的第二语音编码信息；
所述接收模块510，还用于接收所述主叫终端或者所述被叫终端发送的语音通话信息；
转换模块530，用于根据所述第一语音编码信息和所述第二语音编码信息，将所述语音通话信息转换为另一终端所支持的语音通话信息；
发送模块540，用于发送所述转换模块530转换后的所述语音通话信息至所述另一终端。
综上所述，本实施例提供的语音通话装置，通过在接收到主叫终端发送的语音呼叫请求之后，获取主叫终端的第一语音编码信息以及被叫终端的第二语音编码信息，进而在此后接收到主叫终端或者被叫终端发送的语音通话信息时，根据该第一语音编码信息和第二语音编码信息将该语音通话信息转换为另一终端所支持的语音通话信息，发送转换后的语音通话信息至另一终端；解决了相关技术中当出现新的编码类型时，在对语音通话客户端更新期间，语音通话客户端无法进行语音通话的问题；达到了后台服务器可以根据通话两端的语音编码信息直接进行转码，而无需对语音通话客户端进行更新，消除在出现新的编码类型时语音通话客户端可能不能进行语音通话的效果。
基于上述实施例提供的语音通话装置，可选的，所述获取模块520，还用于提取所述语音呼叫请求中携带的所述第一语音编码信息。
可选的，所述获取模块520，还用于根据所述语音呼叫请求中的所述被叫终端的标识，从所述被叫终端所对应的运营商处获取所述第二语音编码信息。
可选的，所述装置用于后台服务器中，所述后台服务器包括：实时传输协议RTP模块和转码模块；
所述获取模块520，还用于：
通过所述RTP模块获取所述第一语音编码信息以及所述第二语音编码信息，发送所述第一语音编码信息以及所述第二语音编码信息至所述转码模块；
通过所述转码模块反馈标识信息至所述RTP模块，所述标识信息用于唯一标识所述第一语音编码信息和所述第二语音编码信息之间的对应关系；
所述接收模块510，还用于通过所述RTP模块接收所述语音通话信息；
所述转码模块530，还用于：
通过所述RTP模块发送所述语音通话信息以及所述标识信息至所述转码 模块；
通过所述转码模块根据所述标识信息，将所述语音通话信息转换为所述另一终端所支持的语音通话信息。
可选的，所述发送模块540，还用于在通话结束之后，通过所述RTP服务器发送通话结束指令至所述转码服务器，所述通话结束指令中包含所述标识信息；
所述装置还包括：
删除模块，用于通过所述转码服务器删除所述标识信息所对应的所述第一语音编码信息和所述第二语音编码信息。
可选的，所述接收模块510，还用于在语音通话过程中，接收目标终端发送的编码信息更新请求，所述目标终端为所述主叫终端或者所述被叫终端，所述编码信息更新请求中携带有更新后的语音编码信息；
所述装置还包括：
更新模块，用于根据更新后的所述语音编码信息更新所述目标终端所对应的语音编码信息。
需要说明的是，本实施例中的RTP模块可以形成为RTP服务器，转码模块可以形成为转码服务器，本实施例对此并不做限定。
需要说明的是，上述实施例提供的语音通话装置，仅以上述各功能模块的划分进行举例说明，实际应用中，可以根据需要而将上述功能分配由不同的功能模块完成，即将设备的内部结构划分成不同的功能模块，以完成以上描述的全部或者部分功能。另外，上述实施例提供的语音通话装置与语音通话方法的方法实施例属于同一构思，其具体实现过程详见方法实施例，这里不再赘述。
在示例性实施例中，还提供了一种包括指令的非临时性计算机可读存储介质，例如包括指令的存储器，上述指令可由服务器中的处理器执行以完成上述语音通话方法。例如，所述非临时性计算机可读存储介质可以是ROM、随机存取存储器(RAM)、CD-ROM、磁带、软盘和光数据存储设备等。
请参考图6，其示出了本申请一个实施例提供的服务器的结构示意图。该服务器用于实施上述实施例中提供的语音通话方法。具体来讲：
所述服务器600包括中央处理单元(CPU)601、包括随机存取存储器(RAM)602和只读存储器(ROM)603的系统存储器604，以及连接系统存储器604和中央处理单元601的系统总线605。所述服务器600还包括帮助计算机内的各个器件之间传输信息的基本输入/输出系统(I/O系统)606，和用于存储操作系统613、应用程序614和其他程序模块615的大容量存储设备607。
所述基本输入/输出系统606包括有用于显示信息的显示器608和用于用户输入信息的诸如鼠标、键盘之类的输入设备609。其中所述显示器608和输入设备609都通过连接到系统总线605的输入输出控制器610连接到中央处理单元601。所述基本输入/输出系统606还可以包括输入输出控制器610以用于接收和处理来自键盘、鼠标、或电子触控笔等多个其他设备的输入。类似地，输入输出控制器610还提供输出到显示屏、打印机或其他类型的输出设备。
所述大容量存储设备607通过连接到系统总线605的大容量存储控制器(未示出)连接到中央处理单元601。所述大容量存储设备607及其相关联的计算机可读介质为服务器600提供非易失性存储。也就是说，所述大容量存储设备607可以包括诸如硬盘或者CD-ROM驱动器之类的计算机可读介质(未示出)。
不失一般性，所述计算机可读介质可以包括计算机存储介质和通信介质。计算机存储介质包括以用于存储诸如计算机可读指令、数据结构、程序模块或其他数据等信息的任何方法或技术实现的易失性和非易失性、可移动和不可移动介质。计算机存储介质包括RAM、ROM、EPROM、EEPROM、闪存或其他固态存储其技术，CD-ROM、DVD或其他光学存储、磁带盒、磁带、磁盘存储或其他磁性存储设备。当然，本领域技术人员可知所述计算机存储介质不局限于上述几种。上述的系统存储器604和大容量存储设备607可以统称为存储器。
根据本申请的各种实施例，所述服务器600还可以通过诸如因特网等网络连接到网络上的远程计算机运行。也即服务器600可以通过连接在所述系统总线605上的网络接口单元611连接到网络612，或者说，也可以使用网络接口单元611来连接到其他类型的网络或远程计算机系统(未示出)。
所述存储器还包括一个或者一个以上的程序，所述一个或者一个以上程序存储于存储器中，且经配置以由一个或者一个以上处理器执行。上述一个或者一个以上程序包含用于执行上述服务器侧的语音通话方法的指令。
应当理解的是，在本文中使用的，除非上下文清楚地支持例外情况，单数形式“一个”(“a”、“an”、“the”)旨在也包括复数形式。还应当理解的是，在本文中使用的“和/或”是指包括一个或者一个以上相关联地列出的项目的任意和所有可能组合。
上述本申请实施例序号仅仅为了描述，不代表实施例的优劣。
本领域普通技术人员可以理解实现上述实施例的全部或部分步骤可以通过硬件来完成，也可以通过程序来指令相关的硬件完成，所述的程序可以存储于一种计算机可读存储介质中，上述提到的存储介质可以是只读存储器，磁盘或光盘等。
以上所述仅为本申请中的部分实施例，并不用以限制本申请，凡在本申请的精神和原则之内，所作的任何修改、等同替换、改进等，均应包含在本申请的保护范围之内。

Claims (18)

  1. 一种语音通话方法，其特征在于，所述方法包括：
    接收主叫终端发送的语音呼叫请求，所述语音呼叫请求中携带有被叫终端的标识；
    获取所述主叫终端的第一语音编码信息，以及，所述被叫终端的第二语音编码信息；
    接收所述主叫终端或者所述被叫终端发送的语音通话信息；
    根据所述第一语音编码信息和所述第二语音编码信息，将所述语音通话信息转换为另一终端所支持的语音通话信息；
    发送转换后的所述语音通话信息至所述另一终端。
  2. 根据权利要求1所述的方法，其特征在于，所述获取所述主叫终端的第一语音编码信息，包括：
    提取所述语音呼叫请求中携带的所述第一语音编码信息。
  3. 根据权利要求1所述的方法，其特征在于，所述获取所述被叫终端的第二语音编码信息，包括：
    根据所述语音呼叫请求中的所述被叫终端的标识，从所述被叫终端所对应的运营商处获取所述第二语音编码信息。
  4. 根据权利要求1所述的方法，其特征在于，所述方法用于后台服务器中，所述后台服务器包括：实时传输协议RTP服务器和转码服务器；
    所述获取所述主叫终端的第一语音编码信息，以及，所述被叫终端的第二语音编码信息，包括：
    通过所述RTP服务器获取所述第一语音编码信息以及所述第二语音编码信息，发送所述第一语音编码信息以及所述第二语音编码信息至所述转码服务器；
    通过所述转码服务器反馈标识信息至所述RTP服务器，所述标识信息用于唯一标识所述第一语音编码信息和所述第二语音编码信息之间的对应关系；
    所述接收所述主叫终端或者所述被叫终端发送的语音通话信息，包括：
    通过所述RTP服务器接收所述语音通话信息；
    所述根据所述第一语音编码信息和所述第二语音编码信息，将所述语音通话信息转换为另一终端所支持的语音通话信息，包括：
    通过所述RTP服务器发送所述语音通话信息以及所述标识信息至所述转码服务器；
    通过所述转码服务器根据所述标识信息，将所述语音通话信息转换为所述另一终端所支持的语音通话信息。
  5. 根据权利要求4所述的方法，其特征在于，所述方法还包括：
    在通话结束之后，通过所述RTP服务器发送通话结束指令至所述转码服务器，所述通话结束指令中包含所述标识信息；
    通过所述转码服务器删除所述标识信息所对应的所述第一语音编码信息和所述第二语音编码信息。
  6. 根据权利要求1至5任一所述的方法，其特征在于，所述方法还包括：
    在语音通话过程中，接收目标终端发送的编码信息更新请求，所述目标终端为所述主叫终端或者所述被叫终端，所述编码信息更新请求中携带有更新后的语音编码信息；
    根据更新后的所述语音编码信息更新所述目标终端所对应的语音编码信息。
  7. 一种语音通话装置，其特征在于，所述装置包括：
    一个或多个处理器；和
    存储器；
    所述存储器存储有一个或多个程序，所述一个或多个程序被配置成由所述一个或多个处理器执行，所述一个或多个程序包含用于进行以下操作的指令：
    接收主叫终端发送的语音呼叫请求，所述语音呼叫请求中携带有被叫终端的标识；
    获取所述主叫终端的第一语音编码信息，以及，所述被叫终端的第二语音编码信息；
    接收所述主叫终端或者所述被叫终端发送的语音通话信息；
    根据所述第一语音编码信息和所述第二语音编码信息，将所述语音通话信 息转换为另一终端所支持的语音通话信息；
    发送转换后的所述语音通话信息至所述另一终端。
  8. 根据权利要求7所述的装置，其特征在于，所述一个或多个程序还包含用于进行以下操作的指令：
    提取所述语音呼叫请求中携带的所述第一语音编码信息。
  9. 根据权利要求7所述的装置，其特征在于，所述一个或多个程序还包含用于进行以下操作的指令：
    根据所述语音呼叫请求中的所述被叫终端的标识，从所述被叫终端所对应的运营商处获取所述第二语音编码信息。
  10. 根据权利要求7所述的装置，其特征在于，所述装置用于后台服务器中，所述后台服务器包括：实时传输协议RTP服务器和转码服务器；所述一个或多个程序还包含用于进行以下操作的指令：
    通过所述RTP服务器获取所述第一语音编码信息以及所述第二语音编码信息，发送所述第一语音编码信息以及所述第二语音编码信息至所述转码服务器；
    通过所述转码服务器反馈标识信息至所述RTP服务器，所述标识信息用于唯一标识所述第一语音编码信息和所述第二语音编码信息之间的对应关系；
    通过所述RTP服务器接收所述语音通话信息；
    通过所述RTP服务器发送所述语音通话信息以及所述标识信息至所述转码服务器；
    通过所述转码服务器根据所述标识信息，将所述语音通话信息转换为所述另一终端所支持的语音通话信息。
  11. 根据权利要求10所述的装置，其特征在于，所述一个或多个程序还包含用于进行以下操作的指令：
    在通话结束之后，通过所述RTP服务器发送通话结束指令至所述转码服务器，所述通话结束指令中包含所述标识信息；
    通过所述转码服务器删除所述标识信息所对应的所述第一语音编码信息和所述第二语音编码信息。
  12. 根据权利要求7至11任一所述的装置，其特征在于，所述一个或多个程序还包含用于进行以下操作的指令：
    在语音通话过程中，接收目标终端发送的编码信息更新请求，所述目标终端为所述主叫终端或者所述被叫终端，所述编码信息更新请求中携带有更新后的语音编码信息；
    根据更新后的所述语音编码信息更新所述目标终端所对应的语音编码信息。
  13. 一种语音通话装置，其特征在于，所述装置包括：
    接收模块，用于接收主叫终端发送的语音呼叫请求，所述语音呼叫请求中携带有被叫终端的标识；
    获取模块，用于获取所述主叫终端的第一语音编码信息，以及，所述被叫终端的第二语音编码信息；
    所述接收模块，还用于接收所述主叫终端或者所述被叫终端发送的语音通话信息；
    转换模块，用于根据所述第一语音编码信息和所述第二语音编码信息，将所述语音通话信息转换为另一终端所支持的语音通话信息；
    发送模块，用于发送所述转换模块转换后的所述语音通话信息至所述另一终端。
  14. 根据权利要求13所述的装置，其特征在于，
    所述获取模块，还用于提取所述语音呼叫请求中携带的所述第一语音编码信息。
  15. 根据权利要求13所述的装置，其特征在于，
    所述获取模块，还用于根据所述语音呼叫请求中的所述被叫终端的标识，从所述被叫终端所对应的运营商处获取所述第二语音编码信息。
  16. 根据权利要求13所述的装置，其特征在于，所述装置用于后台服务器中，所述后台服务器包括：实时传输协议RTP服务器和转码服务器；
    所述获取模块，还用于：
    通过所述RTP服务器获取所述第一语音编码信息以及所述第二语音编码信息，发送所述第一语音编码信息以及所述第二语音编码信息至所述转码服务器；
    通过所述转码服务器反馈标识信息至所述RTP服务器，所述标识信息用于唯一标识所述第一语音编码信息和所述第二语音编码信息之间的对应关系；
    所述接收模块，还用于通过所述RTP服务器接收所述语音通话信息；
    所述转换模块，还用于：
    通过所述RTP服务器发送所述语音通话信息以及所述标识信息至所述转码服务器；
    通过所述转码服务器根据所述标识信息，将所述语音通话信息转换为所述另一终端所支持的语音通话信息。
  17. 根据权利要求16所述的装置，其特征在于，
    所述发送模块，还用于在通话结束之后，通过所述RTP服务器发送通话结束指令至所述转码服务器，所述通话结束指令中包含所述标识信息；
    所述装置还包括：
    删除模块，用于通过所述转码服务器删除所述标识信息所对应的所述第一语音编码信息和所述第二语音编码信息。
  18. 根据权利要求13至17任一所述的装置，其特征在于，
    所述接收模块，还用于在语音通话过程中，接收目标终端发送的编码信息更新请求，所述目标终端为所述主叫终端或者所述被叫终端，所述编码信息更新请求中携带有更新后的语音编码信息；
    所述装置还包括：
    更新模块，用于根据更新后的所述语音编码信息更新所述目标终端所对应的语音编码信息。
PCT/CN2017/087317 2016-07-08 2017-06-06 语音通话方法及装置 WO2018006678A1 (zh)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CN201610539161.7A CN106128468B (zh) 2016-07-08 2016-07-08 语音通话方法及装置
CN201610539161.7 2016-07-08

Publications (1)

Publication Number Publication Date
WO2018006678A1 true WO2018006678A1 (zh) 2018-01-11

Family

ID=57283682

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/CN2017/087317 WO2018006678A1 (zh) 2016-07-08 2017-06-06 语音通话方法及装置

Country Status (2)

Country Link
CN (1) CN106128468B (zh)
WO (1) WO2018006678A1 (zh)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN113923065B (zh) * 2021-09-06 2023-11-24 贵阳语玩科技有限公司 基于聊天室音频的跨版本通信方法、系统、介质及服务器

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN106128468B (zh) * 2016-07-08 2021-02-12 腾讯科技（深圳）有限公司 语音通话方法及装置
CN108986828B (zh) * 2018-08-31 2021-05-28 北京中兴高达通信技术有限公司 呼叫的建立方法及装置、存储介质、电子装置
CN114760273A (zh) * 2022-04-14 2022-07-15 深圳震有科技股份有限公司 语音转发方法、系统、服务器及存储介质

Citations (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6529602B1 (en) * 1997-08-19 2003-03-04 Walker Digital, Llc Method and apparatus for the secure storage of audio signals
CN1937663A (zh) * 2006-09-30 2007-03-28 华为技术有限公司 实现变声通话业务的方法、系统及装置
US20080310612A1 (en) * 2007-06-15 2008-12-18 Sony Ericsson Mobile Communications Ab System, method and device supporting delivery of device-specific data objects
CN103581129A (zh) * 2012-07-30 2014-02-12 中兴通讯股份有限公司 通话处理方法及装置
CN104125138A (zh) * 2013-04-28 2014-10-29 腾讯科技（深圳）有限公司 一种语音通讯方法及装置、系统
CN104580166A (zh) * 2014-12-19 2015-04-29 大唐移动通信设备有限公司 一种基于cscf媒体编码格式转换的方法和装置
CN104994245A (zh) * 2015-05-08 2015-10-21 小米科技有限责任公司 通话实现方法及装置
CN106128468A (zh) * 2016-07-08 2016-11-16 腾讯科技（深圳）有限公司 语音通话方法及装置

Family Cites Families (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN101547343A (zh) * 2009-03-06 2009-09-30 深圳市融创天下科技发展有限公司 一种远程视频监控的系统及方法
JP2011166660A (ja) * 2010-02-15 2011-08-25 Nec Access Technica Ltd 音声記録装置、音声記録方法及び音声記録プログラム
CN103916678A (zh) * 2012-12-31 2014-07-09 中国移动通信集团广东有限公司 多媒体数据转码方法、转码设备及多媒体数据播放系统
CN103414697B (zh) * 2013-07-22 2017-04-05 中国联合网络通信集团有限公司 一种voip自适应语音编码方法、系统及sip服务器
CN103428284A (zh) * 2013-08-07 2013-12-04 合肥迈腾信息科技有限公司 基于云技术的车载网络通话方法
CN105374359B (zh) * 2014-08-29 2019-05-17 中国电信股份有限公司 语音数据的编码方法和系统
CN105491044A (zh) * 2015-12-11 2016-04-13 中青冠岳科技(北京)有限公司 一种基于移动终端进行即时语音通讯的方法和装置

Patent Citations (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6529602B1 (en) * 1997-08-19 2003-03-04 Walker Digital, Llc Method and apparatus for the secure storage of audio signals
CN1937663A (zh) * 2006-09-30 2007-03-28 华为技术有限公司 实现变声通话业务的方法、系统及装置
US20080310612A1 (en) * 2007-06-15 2008-12-18 Sony Ericsson Mobile Communications Ab System, method and device supporting delivery of device-specific data objects
CN103581129A (zh) * 2012-07-30 2014-02-12 中兴通讯股份有限公司 通话处理方法及装置
CN104125138A (zh) * 2013-04-28 2014-10-29 腾讯科技（深圳）有限公司 一种语音通讯方法及装置、系统
CN104580166A (zh) * 2014-12-19 2015-04-29 大唐移动通信设备有限公司 一种基于cscf媒体编码格式转换的方法和装置
CN104994245A (zh) * 2015-05-08 2015-10-21 小米科技有限责任公司 通话实现方法及装置
CN106128468A (zh) * 2016-07-08 2016-11-16 腾讯科技（深圳）有限公司 语音通话方法及装置

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN113923065B (zh) * 2021-09-06 2023-11-24 贵阳语玩科技有限公司 基于聊天室音频的跨版本通信方法、系统、介质及服务器

Also Published As

Publication number Publication date
CN106128468B (zh) 2021-02-12
CN106128468A (zh) 2016-11-16

Similar Documents

Publication Publication Date Title
CN107682657B (zh) 一种基于WebRTC的多人语音视频通话方法及系统
US9716731B2 (en) Consolidated peer-to-peer media sessions for audio and/or video communications
US9210198B2 (en) Method and apparatus for transferring active communication session streams between devices
US20140293997A1 (en) Method, Apparatus, and System for Implementing VOIP Call in Cloud Computing Environment
WO2018006678A1 (zh) 语音通话方法及装置
US10313407B2 (en) Method and apparatus for establishing a session between a thin client and a media gateway for media data streaming
US20130282820A1 (en) Method and System for an Optimized Multimedia Communications System
CN109802913B (zh) 融合会议实现方法及装置、电子设备、可读存储介质
US9894128B2 (en) Selective transcoding
CN112953925B (zh) 基于sip协议和rtc网络实时音视频通信系统及方法
CN108881149B (zh) 一种可视电话设备的接入方法和系统
US20160119468A1 (en) Method and system for rapid internet protocol (ip) communication session setup using interactive push notifications
EP2863591A1 (en) Transmission method for media data stream and thin client
CN112751827B (zh) 一种sip多方会话在宽带集群中的应用方法及系统
US11070665B2 (en) Voice over internet protocol processing method and related network device
US20160036864A1 (en) Providing external application services with an existing private branch exchange media server
CN103997491A (zh) 一种量子保密通信电话用户终端扩展网关系统
CN102932566B (zh) 虚拟桌面基础架构vdi环境下voip电话通话中减少语音失真的方法
US11178006B2 (en) Replacement of collaboration endpoints
JP6183881B2 (ja) コーデック変換ゲートウェイ、コーデック変換方法、及び、コーデック変換プログラム
TWM395968U (en) Communication system encompassing audio platform
CN106453265B (zh) Ip呼叫的调度方法及调度系统、ippbx、服务器
CN105516123A (zh) 网络电话与电话网电话通信的方法及落地电话业务服务器
CN117596231A (zh) 通信方法、终端设备、系统及介质
CN117939068A (zh) 一种支持双向视频通话的电话监控方法和装置

Legal Events

Date Code Title Description
121 Ep: the epo has been informed by wipo that ep was designated in this application

Ref document number: 17823490

Country of ref document: EP

Kind code of ref document: A1

NENP Non-entry into the national phase

Ref country code: DE

122 Ep: pct application non-entry in european phase

Ref document number: 17823490

Country of ref document: EP

Kind code of ref document: A1