WO2017129129A1 - 即时通话方法、装置和系统 - Google Patents

即时通话方法、装置和系统 Download PDF

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Publication number
WO2017129129A1
WO2017129129A1 PCT/CN2017/072591 CN2017072591W WO2017129129A1 WO 2017129129 A1 WO2017129129 A1 WO 2017129129A1 CN 2017072591 W CN2017072591 W CN 2017072591W WO 2017129129 A1 WO2017129129 A1 WO 2017129129A1
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WO
WIPO (PCT)
Prior art keywords
telephone terminal
terminal
data stream
voice call
party voice
Prior art date
Application number
PCT/CN2017/072591
Other languages
English (en)
French (fr)
Inventor
王新亮
李斌
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.)
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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
Priority to US15/759,301 priority Critical patent/US10798138B2/en
Publication of WO2017129129A1 publication Critical patent/WO2017129129A1/zh

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/40—Support for services or applications
    • H04L65/403—Arrangements for multi-party communication, e.g. for conferences
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10—Architectures or entities
    • H04L65/102—Gateways
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10—Architectures or entities
    • H04L65/102—Gateways
    • H04L65/1023—Media gateways
    • H04L65/103—Media gateways in the network
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1069—Session establishment or de-establishment
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H04L65/75—Media network packet handling

Definitions

  • the present application relates to the field of mobile internet application technologies, and in particular, to an instant call method, apparatus, and system.
  • VoIP applications With the increasing use of VoIP applications, users can communicate with each other through a telephone network in cooperation with a telephone terminal, and can also implement VoIP applications through terminal devices capable of accessing the Internet. Both sides, even multi-party voice calls.
  • the telephone network and the Internet are independent of each other, for example, the telephone network is operated by an operator, and the Internet is provided by an Internet service provider. Therefore, the telephone network and the Internet are irrelevant and cannot be used for a telephone terminal. Interworking with the VoIP terminal makes it impossible to implement multi-party voice calls involving the telephone terminal and the VoIP terminal.
  • An object of the present application is to solve the defect that the network telephone terminal and the telephone terminal cannot communicate with each other in the prior art, and the multi-party voice call involved by the telephone terminal and the network telephone terminal cannot be realized.
  • Another object of the present application is to provide an instant call method, apparatus and system capable of interworking between a network telephone terminal and a telephone terminal, and implementing multiple parties involved by the telephone terminal and the network telephone terminal Voice calls.
  • An instant call method the method is performed by a calling party and the calling side is a network phone Terminal, the method includes:
  • the multi-party voice call is performed, where
  • the telephone terminal In the data channel establishment of the terminal, the telephone terminal establishes a data channel corresponding to the remote access address by using a gateway device.
  • An instant call method the method being performed by a network telephone terminal on a called side, the method comprising:
  • the terminal that specifies the multi-party voice call in the multi-party voice call request includes the telephone terminal;
  • the multi-party voice call After the terminal specified by the multi-party voice call request establishes a data channel according to the remote access address, the multi-party voice call is entered.
  • An instant call method the method being performed by a signaling server, the method comprising:
  • the remote access address is returned to the network telephone terminal included in the designated terminal and/or the gateway device corresponding to the telephone terminal by the signaling interaction to establish a multi-party voice call.
  • An instant call device the device running on a calling side, and the calling side is a network power
  • the terminal, the device includes:
  • a request initiation module configured to initiate a multi-party voice call request to the signaling server, where the terminal that specifies the multi-party voice call in the multi-party voice call request includes a telephone terminal;
  • a calling side channel establishing module configured to establish a data channel according to the remote access address returned by the signaling server in response to the multi-party voice call request;
  • a call execution module configured to perform the multi-party voice call after the terminal to be designated establishes a data channel according to the remote access address
  • the telephone terminal In the data channel establishment of the terminal, the telephone terminal establishes a data channel corresponding to the remote access address by using a gateway device.
  • An instant call device the device running in a network telephone terminal on the called side, the device comprising:
  • a request response module configured to respond to the multi-party voice call request forwarded by the signaling server, to obtain a remote access address, where the terminal that specifies the multi-party voice call in the multi-party voice call request includes a telephone terminal;
  • a called side channel establishing module configured to establish a data channel of the network telephone terminal according to the remote access address
  • the multi-party voice call After the terminal specified by the multi-party voice call request establishes a data channel according to the remote access address, the multi-party voice call is entered.
  • An instant call system comprising a request receiving module, a call request response module and a call establishing module running in a signaling server, wherein:
  • the request receiving module is configured to receive a multi-party voice call request initiated by the calling side;
  • the call request response module is configured to, in response to the multi-party voice call request, return a remote access address to the calling side, and initiate a signaling interaction of the call establishment to the terminal specified in the multi-party voice call request, where the designation Terminal includes a telephone terminal;
  • the call establishing module is configured to interact with the network included in the designated terminal by using the signaling
  • the network telephone terminal and/or the gateway device corresponding to the telephone terminal returns a remote access address to establish a multi-party voice call.
  • An instant call system includes a terminal, a signaling server, a data server and a gateway device participating in a multi-party voice call, the terminal comprising a network telephone terminal and a telephone terminal, wherein:
  • the remote access address of the data server and initiating a signaling interaction established by the call to the other terminals of the multi-party voice call, and interacting with other networks through the signaling
  • the gateway device corresponding to the telephone terminal and/or the telephone terminal returns a remote access address
  • the network telephone terminal and/or the gateway device corresponding to the telephone terminal establishes a data channel with the data server by using the remote access address to perform a multi-party voice call.
  • 3 is a flow chart of a method for receiving a data stream transmitted by a data channel in a multi-party voice call in a multi-party voice call, and performing an adaptive mixing process;
  • FIG. 4 is a flow chart of a method for adaptively determining a data stream in FIG. 3 and outputting a data stream without performing an adaptive determination;
  • FIG. 5 is a flow chart of an instant call method performed by a network telephone terminal on the called side in one embodiment
  • FIG. 6 is a flow chart of the method for obtaining a remote access address by responding to a multi-party voice call request forwarded by a signaling server in FIG. 5;
  • FIG. 7 is a flow chart of an instant call method performed by a signaling server in one embodiment
  • FIG. 8 is a flowchart of a method for establishing a multi-party voice call by using a signaling interaction to return a remote access address to a gateway device corresponding to a specified terminal and/or a telephone terminal in FIG. 7;
  • FIG. 9 is a flowchart of an instant call method performed by a server corresponding to a remote access address in another embodiment
  • FIG. 10 is a flow chart of a method for controlling a multi-party voice call between a calling side and a designated terminal through a data channel in FIG. 9;
  • FIG. 11 is a flow chart of a method for forwarding a data stream in FIG. 10 to transmit a data stream to a corresponding terminal through a data channel;
  • FIG. 12 is a topological diagram of an implementation of a multi-party voice call in an embodiment
  • Figure 13 is a timing diagram of the multi-party voice call call process of Figure 12;
  • FIG. 14 is a schematic structural diagram of an instant call device of a network telephone terminal running on a calling side in an embodiment
  • 15 is a schematic structural diagram of an instant messaging device of a network telephone terminal running on a called side in an embodiment
  • 16 is a schematic structural diagram of a call receiving module in an embodiment
  • 17 is a schematic structural diagram of an instant messaging device of a network telephone terminal operating on the called side in an embodiment
  • FIG. 18 is a schematic structural diagram of a request response module in FIG. 17;
  • 19 is a schematic structural diagram of a signaling server in an instant call system in an embodiment
  • FIG. 20 is a schematic structural diagram of a call setup module in FIG. 19;
  • 21 is a schematic structural diagram of a server corresponding to a remote access address in an embodiment
  • FIG. 22 is a schematic structural diagram of a call control module of FIG. 21;
  • FIG. 23 is a schematic structural diagram of a forwarding processing unit of FIG. 22;
  • FIG. 24 is a schematic structural diagram of an adaptive mix forwarding subunit of FIG. 23;
  • FIG. 25 is a schematic structural diagram of a computer system for implementing an instant call method in an embodiment of the present application.
  • the telephone network and the internetwork are independent of each other and irrelevant, so the telephone terminals relying on the telephone network and the network telephone terminals relying on the internetwork will not be able to interconnect on the network.
  • the call of the telephone terminal includes the transmission path, the protocol, the data implementation and the communication interface all follow certain standards, and therefore, the telephone network terminal has good interconnection and interoperability.
  • the VoIP terminal performs call setup and data implementation through various customized protocols in the VoIP application in which it runs, so that the VoIP terminal can make a call through the running VoIP application.
  • an instant call method will be provided to realize interworking between the network telephone terminal and the telephone terminal, thereby implementing multi-party voice calls in which the network telephone terminal and the telephone terminal participate.
  • the calling side performs and the calling side is a network telephone terminal.
  • the instant call method is as shown in FIG. 1 and includes steps S110-S150.
  • Step S110 Initiating a multi-party voice call request to the signaling server, and the terminal that specifies the multi-party voice call in the multi-party voice call request includes the telephone terminal.
  • any one of the network telephone terminals can be used as the calling side.
  • the multi-party voice call request can be initiated to the signaling server.
  • the terminal that participates in the multi-party voice call is designated as the called side, and includes any one or any combination of the telephone terminals, and may also include one or more network telephone terminals.
  • Multi-party voice call refers to a call process in which two types of terminals, a network telephone terminal and a telephone terminal, are combined, and during the call, the network telephone terminal makes a call with the telephone terminal through the network telephone application running by itself.
  • the user specifies a number of terminals as the called side to participate in the upcoming multi-party voice call by specifying a number of number identifiers in the running VoIP application.
  • the process of specifying the number identification in the VoIP application may be a dialing process.
  • the signaling server acts as a remote server that cooperates with the terminal to implement signaling interaction to control establishment of a multi-party voice call.
  • the multi-party voice call request initiated to the signaling server includes the calling side network telephone terminal.
  • Terminal system type includes a calling number identifier and a called number identifier.
  • the service for initiating the multi-party voice call can be obtained, so that multiple applications in the network phone terminal can support the multi-party voice call function, and the multi-party voice call is applied to each service scenario.
  • Step S130 Establish a data channel according to the remote access address returned by the signaling server in response to the multi-party voice call request.
  • the signaling server After receiving the multi-party voice call request, the signaling server will return the remote access address to the network telephone terminal accordingly.
  • the remote access address returned is the address of a server, and the server is used to control the transmission of the data stream in the multi-party voice call.
  • the server may be the same server as the signaling server, or may be other servers different from the signaling server.
  • the specific server and signaling server layout may be flexibly configured according to actual needs.
  • the network telephone terminal that initiates the multi-party voice call request directly establishes a data channel with the server according to the remote access address, and the data channel is used to implement transmission of the data stream related to the network telephone terminal.
  • the data stream generated by the network telephone terminal transmitting the input to the server may be included, or the server may return the data stream from the terminal to the network telephone terminal.
  • the function of the signaling server is to access the multi-party voice call for each terminal, and the server corresponding to the remote access address serves to provide a relay for the data flow generated between the terminals in the multi-party voice call.
  • Step S150 The terminal to be designated performs a multi-party voice call after establishing a data channel according to the remote access address.
  • the telephone terminal During the establishment of the data channel of the terminal, the telephone terminal establishes a data channel corresponding to the remote access address through the gateway device.
  • the network telephone terminal on the calling side will also wait for a preset time to wait for the specified terminal to establish a data channel in response to the multi-party voice call request before entering the multi-party voice channel.
  • the specified one or several terminals may complete the establishment of the data channel, or all the designated terminals may be completed.
  • the establishment of the data channel regardless of the circumstances, can be initiated Multi-party voice call.
  • any one or any combination of terminals can participate in the multi-party voice call initiated by the network telephone terminal on the calling side as long as the data channel is established. in.
  • the called side for the network telephone terminal, can directly establish a data channel according to the remote access address and the corresponding server; for the telephone terminal, since the telephone network and the internet network are independent of each other, the gateway device is required as the telephone network and A border device between the Internet gateways, the gateway device establishes a data channel of the telephone terminal with the corresponding server according to the remote access address, so that the telephone terminal can establish a connection with the server through the gateway device.
  • Telephone terminals include landlines and mobile phones.
  • the gateway device is a Voip gateway, one side of which is connected to the telephone network, and the other side is connected to the Internet to implement protocol and data conversion between the two networks, thereby implementing interworking between the two networks. .
  • the telephone network and the internet network can be interconnected, thereby realizing the conversation between the network telephone terminal and the telephone terminal, and even with the assistance of the network telephone terminal, the telephone terminal can also participate in the network telephone.
  • the telephone terminal In the multi-party voice call performed by the terminal, the boundaries and restrictions of the call are released.
  • the method as described above, as shown in FIG. 2 further includes steps S210 and S230.
  • each network telephone terminal participating in the multi-party voice call is configured to respond to the multi-party voice call request forwarded by the signaling server to obtain a remote access address.
  • each network telephone terminal that participates in the multi-party voice call is the called side, and correspondingly, the forwarded multi-party voice call request will be used as a call request for the network telephone terminal on the called side to be on the called side.
  • a ringing notification is made in the VoIP terminal and waiting for a response.
  • the network telephone terminal will ring after receiving the multi-party voice call request to notify the initiated multi-party voice call call.
  • the response to the ringing is a response of the multi-party voice call request, which includes answering, rejecting, and timeout.
  • the network telephone terminal detects the answering operation, it responds to obtain the far end returned by the signaling server. address.
  • Step S230 establishing a data channel of the network telephone terminal according to the remote access address.
  • each network telephone terminal that participates in the multi-party voice call is designated to obtain the remote access address
  • the respective data channels are established according to the remote access address, thereby establishing the call.
  • any network telephone terminal on the called side can participate in the voice call between the calling side and the telephone terminal by answering, thereby making the voice call strong. Flexibility.
  • the method as described above further comprises: in the multi-party voice call, the network telephone terminal receives the data stream transmitted by the data channel, and performs adaptive mixing processing and then plays.
  • any network telephone terminal will receive the data stream, wherein the received data stream may be one data stream or multiple data streams. Therefore, adaptive mixing processing is required.
  • the received data stream may be a data stream, for example, only one terminal currently performs voice input. However, there may be multiple data streams, that is, some or all of the other terminals participating in the voice call perform voice input.
  • the multi-channel data stream mixing involved in the multi-party voice call is adaptively and dynamically switched locally at the remote end or the network telephone terminal. Therefore, if the far-end has been mixed, the network telephone terminal The received data stream is necessarily a data stream; if the far end is not mixed, the data stream received by the network telephone terminal is a multi-channel data stream, and the multi-channel data stream needs to be mixed.
  • any network telephone terminal can perform playback after completing the adaptive mixing processing of the data stream to complete the voice output in the multi-party voice call.
  • the adaptive mixing process includes an adaptive judging process of whether the data stream is mixed or not, and an execution process based on the adaptive judging result.
  • the step of the network telephone terminal receiving the data stream transmitted by the data channel in the multi-party voice call and performing the adaptive mixing process and the playing includes the steps S310-350.
  • Step S310 in the multi-party voice call, each network telephone terminal receives the data stream through the data channel, and the data stream is from other terminals.
  • a multi-party voice call is a call between a plurality of terminals including a network telephone terminal and a telephone terminal, and both establish respective data channels.
  • the network telephone terminal receives the data stream through its own established data channel to learn the voice input performed by other terminals.
  • Step S330 adaptively determining the data stream, and outputting an adaptive determination result that the data stream is not mixed.
  • the identity information identifies the terminal from which the data stream comes from.
  • the data stream that is mixed by the server also identifies the target terminal corresponding to the data stream by using the identity identification information, and the target terminal is the terminal that receives the data stream.
  • any one of the data streams is accompanied by the identity information of the terminal from which the tag is sent; in the data stream obtained by the server mixing, except for the attached identifier
  • the identity information of the corresponding target terminal is also attached.
  • the adaptive determination will be based on the attached identity information.
  • Step S350 performing a mixing process of the data stream according to the adaptive determination result, and playing the data stream obtained by the mixing process.
  • the terminal local mixing is performed at the network telephone terminal, and the received data stream is superimposed into one data stream, and then played to complete the voice in the multi-party voice call. Output.
  • the adaptive judging and mixing processing process as described above provides a concrete implementation for dynamic adaptive switching of server mixing and terminal local mixing, thereby enhancing data implementation in multi-party voice calls. flexibility.
  • step S330 is as shown in FIG. 4, and includes steps S331-335.
  • step S331 it is determined whether the data stream is accompanied by its own identity information. If not, the process proceeds to step S333, and if YES, the process proceeds to step S335.
  • the network telephone terminal that receives the data stream, it is determined whether any of the received data streams is accompanied by the identity information of the network telephone terminal itself, and if so, the data stream is transmitted after the server is mixed. Therefore, the adaptive judgment result that the corresponding output data stream has been mixed is obtained, and the adaptive judgment result that the data stream is not mixed is output.
  • Step S333 outputting an adaptive determination result that the data stream is not mixed.
  • Step S335 outputting an adaptive determination result that the data stream has been mixed, and the adaptive determination result triggers playing of the data stream.
  • the identity information and the number identification are involved in the multi-party voice call, wherein the identity information is used to identify the terminal from the data stream and the target terminal for which the server mixes; the number identification is used for Mark the source terminal and target terminal corresponding to the data stream transmission.
  • the target terminal refers to the same terminal for the same data stream, and the target terminal involved in the number identification and the identity information.
  • the method as described above further comprises: in the multi-party voice call, any network telephone terminal encodes the input call data to obtain the data stream input by itself, and transmits the data stream through the established data channel.
  • any network telephone terminal will perform voice input.
  • the call data obtained by the voice input is encoded to form a data stream input by itself, and the data stream needs to be transmitted to the remote end through the data channel established by itself. Accessing the server corresponding to the address, and then sending the data stream to multiple terminals participating in the multi-party voice call under the control of the server.
  • any network telephone terminal establishes its own data channel, and realizes the data stream transmission through the data channel, thereby realizing the data stream generation and transmission of each network telephone terminal in the multi-party voice call.
  • a network telephone terminal capable of initiating a multi-party voice call and realizing transmission and processing of a data stream in a multi-party voice call is provided, in addition to providing a voice call capable of participating in a multi-party voice call.
  • the VoIP terminal and the telephone terminal access multi-party voice calls in cooperation with the gateway device, thereby forming a multi-party voice communication supported by the Internet and the telephone network.
  • the VoIP terminal and the telephone terminal are allowed to talk freely.
  • the telephone terminal it is a support for multi-party voice call by means of a network telephone terminal, that is to say, through the initiation of a multi-party voice call request in the network telephone terminal, the telephone terminal can also not support multi-party voice in the original.
  • participating in the multi-party voice call initiated by the VoIP terminal greatly expands the function of the telephone terminal.
  • an instant call method is also proposed correspondingly, which is performed by the network telephone terminal on the called side, as shown in FIG. 5, including steps S410-450.
  • Step S410 Acknowledge the multi-party voice call request forwarded by the signaling server to obtain a remote access address, and the terminal that specifies the multi-party voice call in the multi-party voice call request includes the telephone terminal.
  • Step S430 establishing a data channel of the network telephone terminal according to the remote access address.
  • Step S450 The terminal designated by the multi-party voice call request enters the multi-party voice call after establishing the data channel according to the remote access address.
  • the step S410 includes S411-S415.
  • Step S411 Receive a multi-party voice call request forwarded by the signaling server to the calling side.
  • the signaling server is configured to control the establishment of a multi-party voice call. After receiving the multi-party voice call request, the signaling server forwards the called number identifier carried in the multi-party voice call request.
  • Step S413 generating an answer response signaling according to the detected answering operation, and transmitting the answering response signaling to the signaling server.
  • the network telephone terminal receiving the multi-party voice call request will announce the ringing.
  • the current ringing notification is interrupted; if the ringing notification waits for the timeout, the hanging operation is triggered.
  • the current ringing notification is also interrupted, and the triggering of the refusal operation and the hang-up operation will stop the current multi-party voice call establishment process.
  • an answering response signaling is generated and transmitted to the signaling server.
  • Step S415 Receive a remote access address sent by the signaling server according to the answer response signaling.
  • the signaling server After receiving the answering response signaling of the network telephone terminal, the signaling server sends a remote access address to the network telephone terminal, and provides a node capable of relaying the data stream to the multi-party voice call performed by the network telephone terminal through the remote access address, and further The terminal in the multi-party voice call is connected through the remote access address, so that the multi-party voice call can be smoothly performed.
  • an instant call method is also provided accordingly, the method being performed by a signaling server, as shown in FIG. 7, including steps S510-S550.
  • Step S510 receiving a multi-party voice call request initiated by the calling side.
  • the signaling server receives the multi-party voice call request initiated by any network telephone terminal, so that the network telephone terminal that initiates the multi-party voice call request can be known according to the identification number carried in the multi-party voice call request. And VoIP terminals and telephone terminals that are designated to participate in multi-party voice calls.
  • Step S530 in response to the multi-party voice call request, returning the remote access address to the calling side, and initiating the signaling interaction of the call establishment to the terminal specified in the multi-party voice call request, where the designated terminal includes the telephone terminal.
  • the signaling server After receiving the multi-party voice call request, the signaling server should respond to the calling side, and should also initiate a call to the designated terminal through the multi-party voice call request.
  • the signaling server returns a remote access address to the calling side according to the calling number identifier carried in the multi-party voice call request, so as to trigger the data channel establishment of the calling side network telephone terminal.
  • the signaling server initiates a signaling interaction established by the call to the corresponding terminal according to the called number identifier carried in the multi-party voice call request, to trigger the response between the signaling server and the terminal and the data channel establishing process.
  • the gateway device for the response between the signaling server and the designated terminal and the data channel establishment process, if the terminal is a telephone terminal, it is implemented by the gateway device.
  • step S550 the remote access address is returned to the designated terminal and/or the gateway device corresponding to the telephone terminal by signaling interaction to establish a multi-party voice call.
  • the signaling server sends a remote access address to the network telephone terminal on the calling side on the one hand, and the designated terminal includes the network telephone terminal and/or the telephone terminal, and thus also corresponds to the network telephone terminal and/or the telephone terminal.
  • the gateway device sends a remote access address to trigger the network call on the called side. Multi-party voice call setup process for terminals and/or telephone terminals.
  • the establishment of a multi-party voice call between the terminals is implemented by the signaling server as described above, and a multi-party voice call is established for the network telephone terminal and the telephone terminal under the action of the signaling server, thereby implementing a call between the terminals, and the call is implemented.
  • the telephone terminal can be added to the multi-party voice call initiated by the network telephone terminal, and a feasible signaling server is provided for the multi-party voice call of the telephone terminal, so that the signaling server has a very good relationship between the network telephone terminal and the telephone terminal. Good interoperability, free communication between the VoIP terminal and the telephone terminal.
  • the signaling server initiates a signaling interaction of the multi-party voice call establishment between the terminals on the called side, and also identifies the type corresponding to each terminal specified in the multi-party voice call request initiated by the current call to obtain The corresponding terminal identifies the result.
  • the terminal identification result will indicate whether the corresponding terminal on the called side is a network telephone terminal or a telephone terminal, and therefore, a signaling interaction will be initiated according to the terminal identification result.
  • the multi-party voice call request can be directly forwarded to the network telephone terminal, and the telephone terminal needs to be converted into a telephone call request before being transmitted.
  • the identification of the terminal will be based on the called identification number. Specifically, it is determined whether the called identification number is a registered identification number. If yes, the terminal obtains the terminal identification result of the network telephone terminal, and if not, the terminal obtains the terminal identification result of the telephone terminal.
  • any network telephone terminal after logging in, it has already registered its corresponding identification number in the signaling server to maintain the online state in which it is located in the signaling server, so that the call can be made at any time.
  • step S550 includes steps S551-S555.
  • Step S551 the telephone terminal is converted into a multi-party voice call request to obtain a telephone call request, and sent to the corresponding gateway device, and the telephone call request is sent to the telephone terminal through the gateway device.
  • the terminal specified in the multi-party voice call request includes a telephone terminal. Since the protocols and data involved in the internet and the telephone terminal are different, the multi-party voice call request needs to be converted to be converted into a gateway device and the telephone terminal can recognize Phone call request.
  • the signaling server After the signaling server forwards the multi-party voice call request for the telephone terminal to obtain the telephone call request, the signaling server sends the obtained telephone call request to the gateway device, so that the telephone call request can be delivered to the telephone terminal through the gateway device.
  • Step S553 receiving the answer response signaling forwarded by the gateway device, and answering the response signaling is the end of the call.
  • the terminal responds to the telephone call request and returns to the gateway device.
  • the gateway device After receiving the telephone call request sent by the signaling server, the gateway device will ring the telephone terminal by telephone call request to wait for the telephone terminal to answer.
  • the answering response signaling is generated correspondingly and sent to the gateway device to report the response of the telephone terminal to the signaling server through the gateway device.
  • Step S555 in response to the answer response signaling, returning a remote access address to the gateway device to control the gateway device to establish a multi-party voice call through the remote access address.
  • the signaling server learns the answering operation triggered in the telephone terminal, the signaling server establishes a multi-party voice call by returning the remote access address to the gateway device.
  • the network device establishes a data channel for the telephone terminal by using the remote access address transmission performed by the signaling server, so as to complete the establishment of the multi-party voice call on the side of the telephone terminal.
  • the VoIP terminal establishes a data channel directly with the server corresponding to the remote access address, which is different from the VoIP terminal.
  • the telephone terminal establishes a data channel by the gateway device and the server corresponding to the remote access address.
  • the establishment of a data channel between the signaling server and the gateway device is implemented through signaling interaction between the signaling server, the gateway device, and the telephone terminal, thereby enabling the introduction of the telephone terminal into the multi-party voice call.
  • the introduction of the telephone terminal successfully implemented by the signaling server greatly enhances the call performance.
  • the method further includes steps S610 and S630.
  • Step S610 The server corresponding to the remote access address establishes a data channel with the calling side and the designated terminal respectively.
  • each terminal includes a network telephone terminal as a calling side, and also includes a network telephone terminal and/or a telephone terminal as the called side.
  • Step S630 controlling a multi-party voice call between the calling side and the designated terminal through the data channel.
  • the server corresponding to the remote access address receives the data stream, and the data stream is from any terminal of the multi-party voice call. At this time, the data stream is phased.
  • the processing of the off is to enable the terminals of the multi-party voice call to obtain the data stream from the other terminals through the data channel of each terminal, thereby performing multi-party voice calls between the terminals.
  • the server corresponding to the remote access address may be a data server, which is used for data implementation in a multi-party voice call, such as transmission control, etc., which may be mounted on the same server device as the signaling server, or may be respectively installed on different servers. Configure the topology of the server based on actual operations.
  • the step S630 includes steps S631 and S633.
  • Step S631 receiving a data stream sent by any network telephone terminal in the multi-party voice call or a data stream sent by the telephone terminal through the gateway device.
  • the data stream from any network telephone terminal or telephone terminal is transmitted to the server corresponding to the remote access address.
  • Step S633 performing forwarding processing on the data stream, so that the data stream is transmitted to the corresponding terminal through the data channel.
  • the received data stream is one way or multiple channels, and therefore, forwarding processing is required.
  • the forwarding process may be direct forwarding, or may be two operations of mixing and forwarding.
  • the received data stream is a multiplexed data stream
  • step S633 is as shown in FIG. 11, including steps S6331 and S6333.
  • Step S6331 determining a receiving end of the multi-path data stream, where the receiving end comprises a network telephone terminal and/or a telephone terminal.
  • the server corresponding to the remote access address for a multi-party voice call, it will currently receive multiple data streams. At this time, multiple receiving ends corresponding to the multiple data streams are separately processed.
  • the terminal other than the terminal corresponding to the identity identification information is the receiving end.
  • Step S6333 For each receiving end, when the receiving end is a network telephone terminal and supports mixing processing, the data stream corresponding to the receiving end in the multi-channel data stream is forwarded, and the receiving end is a telephone terminal or the receiving end is a network telephone terminal and does not support mixing. During the tone processing, the forwarded data stream is mixed and forwarded.
  • the forwarded data stream is any one of the data streams or any combination of the data streams.
  • Each of the received data streams has a corresponding receiving end. Therefore, each of the determined plurality of receiving ends has a corresponding one channel or a combined number of data streams. Therefore, each corresponding data stream will be processed for each receiving end.
  • the network telephone terminal or the gateway when the network telephone terminal or the gateway completes the data channel establishment and enters the multi-party voice call, it also notifies the server corresponding to the remote access address whether it supports the mixing process. On the basis of this, if a receiving end is a network telephone terminal and supports mixing processing, the corresponding data stream is directly forwarded; if a receiving end is a telephone terminal or the receiving end is a network telephone terminal and does not support mixing processing, then The data stream needs to be mixed, so that the multi-channel data stream can be transmitted to the telephone terminal and the network telephone terminal that does not support the mixing process, thereby ensuring that only the telephone terminal capable of receiving one data stream can perform multi-party voice conversation, and further Compatible with various types of VoIP terminals, increasing the range of applications for multi-party voice calls.
  • the mixing process performed on the telephone terminal further includes encoding of a telephone format
  • the forwarded data stream is a code stream supported by the telephone terminal
  • the method further includes:
  • the gateway device receives the code stream supported by the telephone terminal and delivers the code stream to the corresponding telephone terminal.
  • the server corresponding to the remote access address in addition to mixing the data stream to be transmitted to the telephone terminal, is also encoded so as to be finally forwarded to the gateway device, and then the data stream forwarded by the gateway device to the corresponding telephone terminal is the network.
  • the device and the telephone terminal are identifiable, and thus conform to the protocol and data implementation in the telephone network.
  • step S6333 when the receiving end is a telephone terminal or the receiving end is a network telephone terminal and the mixing process is not supported, the data stream to be forwarded is subjected to mixing processing and then forwarded:
  • the receiving terminal phone terminal or the receiving end is a network telephone terminal and does not support the mixing process
  • the data stream to be forwarded is superimposed to output one data stream, and the identity information of the receiving end is included in the output data stream.
  • the superposition of data streams can be achieved by linear superposition of multiple digital signals in the time domain.
  • the identity information of the receiving end is attached, so that the receiving end can easily and quickly determine whether the server mixing has been performed by using the identity information.
  • the system architecture and implementation of the server and the gateway device are provided for the implementation of the multi-party voice call, thereby enabling the implementation of the network telephone terminal and the telephone terminal.
  • the multi-party voice call includes n terminals, wherein, as shown in FIG. 12, n is an integer not less than 4, and the multi-party voice call is taken as an example for illustration.
  • the n terminals will include at least a network telephone terminal 1, a network telephone terminal 2, a telephone terminal 1, and a telephone terminal 2.
  • the multi-party voice call includes a multi-party voice call process and a data processing process after the multi-party voice call is established.
  • a network telephone terminal 770 acts as a calling side to initiate a multi-party voice call request to the signaling server 710.
  • the other network telephone terminal 770 and the telephone terminal 790 act as the called side, and perform a multi-party voice call request under the control of the signaling server 710.
  • the response that is, the sequence process shown in FIG. 13 is performed, so that each terminal can obtain the remote access address by the signaling server 710, thereby establishing the data channel of each terminal, and completing the multi-party voice call call process.
  • the remote access address is the address of the data server 730.
  • the encoded call data is encoded by itself and forwarded to the data server 730 through the data channel.
  • the voice data entered in any of the telephone terminals 790 is sent to the Voip gateway 750, which is forwarded by the Voip gateway 750 to the data server 730, which immediately forwards this to obtain a data stream that the network telephone terminal 770 can resolve.
  • the data server 730 determines which terminals the multiplexed data stream will be sent to, that is, determines the receiving end; and then forwards it after performing direct forwarding or mixing processing for each receiving end.
  • the data server 30 performs the mixing process to perform the forwarding of the data stream, and the mixing processing further includes a process of encoding in accordance with the format required by the telephone to enable the data stream recognized by the telephone terminal 790.
  • the network telephone terminal 770 as described above may be various mobile terminals and fixed terminals, for example, a smart phone, a tablet phone, a desktop computer, etc., and any terminal capable of accessing the internet network and running the network phone application can function as the network telephone terminal 770.
  • a real-time calling device is also provided, the device is running on the calling side, and the calling side is a network telephone terminal. As shown in FIG. 14, the device includes a request initiating module 110, a calling party. The side channel establishing module 130 and the call execution module 150, wherein:
  • the request initiating module 110 is configured to initiate a multi-party voice call request to the signaling server, and the terminal that specifies the multi-party voice call in the multi-party voice call request includes the telephone terminal.
  • the calling side channel establishing module 130 is configured to establish a data channel according to the remote access address returned by the signaling server in response to the multi-party voice call request.
  • the call execution module 150 is configured to perform a multi-party voice call after the terminal to be designated establishes a data channel according to the remote access address.
  • the telephone terminal During the establishment of the data channel of the terminal, the telephone terminal establishes a data channel corresponding to the remote access address through the gateway device.
  • the apparatus as described above, as shown in FIG. 15, further includes a request response module 210 and a called side channel establishment module 230, and the request response module 210 and the called side channel establishment module 230 operate on the specified participant voice.
  • a request response module 210 and a called side channel establishment module 230 operate on the specified participant voice.
  • the request response module 210 is configured to respond to the multi-party voice call request forwarded by the signaling server to obtain a remote access address.
  • the called side channel establishing module 230 is configured to establish a data channel of the network terminal according to the remote access address.
  • the apparatus as above further includes a call receiving module running on the network telephone terminal in the multi-party voice call, the call receiving module is configured to receive the data stream transmitted by the data channel, and perform adaptive mixing processing and then play .
  • the call receiving module 300 includes a receiving and executing unit 310, an adaptive determining unit 330, and a local mixing unit 350, where:
  • the receiving execution unit 310 is configured to receive the data stream through the data channel, where the data stream is from other terminals.
  • the adaptive determination unit 330 is configured to adaptively determine the data stream, and output an adaptive determination result that the data stream is not mixed.
  • the local mixing unit 350 is configured to perform mixing processing of the data stream according to the adaptive determination result, and play the data stream obtained by the mixing processing.
  • the adaptive determining unit 330 is configured to determine whether the data stream carries its own identity If the information is negative, the output data stream is not subjected to the adaptive determination result of the mixing. If yes, the output data stream is subjected to the adaptive determination result of the mixing, and the adaptive determination result triggers the playback of the data stream.
  • the apparatus as described above further includes a data stream input module, the data stream input module running on any network telephone terminal in the multi-party voice call, for encoding the input call data to obtain the data stream input by itself, and Transmission through established data channels.
  • an instant call device is also provided correspondingly, the device is operated in a network telephone terminal on the called side, and the device comprises a request response module 410 and a called side call establishment module, as shown in FIG. 430.
  • the request response module 410 is configured to respond to the multi-party voice call request forwarded by the signaling server to obtain a remote access address, and the terminal that specifies the multi-party voice call in the multi-party voice call request includes the telephone terminal.
  • the called side call establishing module 430 is configured to establish a data channel of the network telephone terminal according to the remote access address.
  • the terminal designated by the multi-party voice call request enters the multi-party voice call after establishing the data channel according to the remote access address.
  • the request response module 410 includes a forwarding request receiving unit 411, a signaling generating unit 413, and an address receiving unit 415, where:
  • the forwarding request receiving unit 411 is configured to receive a multi-party voice call request forwarded by the signaling server to the calling side.
  • the signaling generating unit 413 is configured to generate an answer response signaling according to the detected answering operation, and transmit the receiving response signaling to the signaling server.
  • the address receiving unit 415 is configured to receive a remote access address that is sent by the signaling server according to the answer response signaling.
  • an instant call system is also provided correspondingly. As shown in FIG. 19, the system includes a request receiving module 510, a call request response module 530 and a call establishing module 550 running in the signaling server, wherein :
  • the request receiving module 510 is configured to receive a multi-party voice call request initiated by the calling side.
  • the call request response module 530 is configured to return a remote access address to the calling side in response to the multi-party voice call request, and initiate a signaling interaction of the call establishment to the terminal specified in the multi-party voice call request, where the designated terminal includes the telephone terminal.
  • the call establishing module 550 is configured to return a remote access address to the gateway device corresponding to the network telephone terminal and/or the telephone terminal included in the designated terminal by signaling interaction to establish a multi-party voice call.
  • the call setup module 550 includes a request conversion unit 551, an acknowledgement signaling receiving unit 553, and a signaling response unit 555, where:
  • the request conversion unit 551 is configured to convert the multi-party voice call request for the telephone terminal to obtain a telephone call request, and send the request to the corresponding gateway device, and the telephone call request is sent to the telephone terminal through the gateway device.
  • the response signaling receiving unit 553 is configured to receive the answer response signaling forwarded by the gateway device, where the answer response signaling is returned by the telephone terminal to the gateway device in response to the telephone call request.
  • the signaling response unit 555 is configured to return a remote access address to the gateway device in response to the answer response signaling, to control the gateway device to establish a multi-party voice call through the remote access address.
  • the system further includes a channel establishment module 610 and a call control module 630.
  • the channel establishment module 610 and the call control module 630 run in a server corresponding to the remote access address.
  • the channel establishing module 610 is configured to establish a data channel with the calling terminal and the designated terminal respectively.
  • the call control module 630 is configured to control a multi-party voice call between the calling side and the designated terminal through the data channel.
  • the call control module 630 includes a data stream receiving unit 631 and a forwarding processing unit 633, where:
  • the data stream receiving unit 631 is configured to receive a data stream sent by any network telephone terminal in the multi-party voice call or a data stream sent by the telephone terminal through the gateway device.
  • the forwarding processing unit 633 is configured to perform line forwarding processing on the data stream, so that the data stream is transmitted to the corresponding terminal through the data channel.
  • the received data stream is a multiplexed data stream
  • the forwarding processing unit 633 includes a receiving end determining subunit 6331 and an adaptive mixing forwarding subunit 6333, as shown in FIG.
  • a receiving end determining subunit 6331 for determining a receiving end of the multiplexed data stream, wherein the receiving The terminal includes a network telephone terminal and/or a telephone terminal.
  • the adaptive mixing and forwarding sub-unit 6333 is configured to forward the data stream corresponding to the receiving end of the multi-channel data stream when the receiving end is the network telephone terminal and supports the mixing process, and when the receiving end is the telephone terminal or the receiving end is the network telephone When the terminal does not support the mixing process, the data stream to be forwarded is mixed and forwarded.
  • the forwarded data stream is any one of the data streams or any combination of the data streams.
  • the mixing processing performed on the telephone terminal includes encoding of a telephone format, and the forwarded data stream is a code stream supported by the telephone terminal, and the system further includes a gateway device, where the gateway device is configured to receive the code stream supported by the telephone terminal, and The code stream is delivered to the corresponding telephone terminal.
  • the adaptive mix forwarding sub-unit 6333 includes a data stream superposition sub-unit 63331 and a forwarding execution sub-unit 63333, wherein:
  • the data stream superposition sub-unit 63331 is configured to: when the receiving end is a telephone terminal or the receiving end is a network telephone terminal and does not support the mixing processing, superimpose the data stream to be forwarded to output one data stream, and in the output one-way data stream The identity information of the receiving end is included;
  • the forwarding execution sub-unit 63333 is configured to forward the one-way data stream with the identity identification information of the receiving end to the receiving end.
  • an instant call system is also provided accordingly.
  • the system includes a terminal participating in a multi-party voice call, a signaling server 710, a data server 730, and a gateway device 750, the terminal including Network telephone terminal 770 and telephone terminal 790, wherein:
  • the gateway device 750 corresponding to the telephone terminal 790 returns the remote access address.
  • the network telephone terminal 770 and/or the gateway device 750 corresponding to the telephone terminal 90 establishes a data channel with the data server 730 through the remote access address to perform a multi-party voice call.
  • An instant call method provided by this embodiment is implemented by a computer program, and the instant call device and system are stored in the computer system 900 as shown in FIG. 25, in the computer system 900. Running in.
  • the computer system 900 includes a processor 910, a memory 920, and a system bus 922.
  • Various system components including memory 920 and processor 910, are coupled to system bus 1022.
  • Processor 910 is a hardware used to execute computer program instructions through basic arithmetic and logic operations in a computer system.
  • the memory 920 is a physical device for temporarily or permanently storing computing programs or data (e.g., program state information), program instructions for storing processing methods of the system telephone during voice calls, and other data, and the processor 910 will execute the memory 920. In the program instructions, and monitor the status change of the system phone, and respond to this, to achieve a temporary interruption or recovery of the voice call.
  • the memory 920 includes a read only memory (ROM) or a flash memory (neither shown), and a random access memory (RAM), which generally refers to a main memory loaded with an operating system and an application.
  • ROM read only memory
  • RAM random access memory
  • Computer system 900 also includes a display interface 930 (eg, a graphics processing unit), a display device 940 (eg, a liquid crystal display), an audio interface 950 (eg, a sound card), and an audio device 960 (eg, a speaker).
  • Display device 940 will be a display for making a voice call interface.
  • Computer system 900 typically includes a storage device 970.
  • Storage device 970 can be selected from a variety of computer readable media, which are any available media that can be accessed by computer system 900, including both mobile and fixed media.
  • a computer readable medium includes, but is not limited to, a flash memory (micro SD card), a CD-ROM, a digital versatile disc (DVD) or other optical disc storage, a magnetic tape cartridge, a magnetic tape, a magnetic disk storage, or other magnetic storage device, or Any other medium that stores the required information and is accessible by computer system 900.
  • Computer system 900 also includes an input device 980 and an input interface 990 (eg, an IO controller).
  • input device 980 such as a keyboard, mouse, touch panel device on display device 940.
  • Input device 980 is typically coupled to system bus 922 via input interface 990, but may be coupled via other interfaces or bus structures, such as a universal serial bus (USB).
  • USB universal serial bus
  • Computer system 900 can be logically coupled to one or more network devices in a network environment.
  • the network device can be a personal computer, server, router, smart phone, tablet or other public network node.
  • Computer system 900 is coupled to the network device via a local area network (LAN) interface 1000 or mobile communication unit 1010.
  • Local area network (LAN) means in a limited area, For example, homes, schools, computer labs, or office buildings that use online media, and computer networks that are interconnected.
  • WiFi and twisted pair cabling Ethernet are the two most commonly used technologies for building LANs.
  • WiFi is a technology that enables computer systems 900 to exchange data or connect to a wireless network via radio waves.
  • the mobile communication unit 1010 can answer and make calls over a radio communication line while moving within a wide geographical area. In addition to the call, the mobile communication unit 1010 also supports Internet access in a 2G, 3G or 4G cellular communication system providing mobile data services.
  • computer system 900 suitable for use in the present application is capable of performing the specified operations of the instant call described above.
  • Computer system 900 performs these operations in the form of program instructions that processor 910 runs in a computer readable medium.
  • These program instructions can be read into memory 920 from storage device 970 or from another device via local area network interface 1000.
  • the program instructions stored in the memory 920 cause the processor 910 to perform the above-described method of implementing image processing.
  • the present application can also be implemented by a hardware circuit or a hardware circuit in combination with a software instruction. Thus, implementation of the application is not limited to any specific combination of hardware circuitry and software.
  • Computer system 900 is merely an example of a computer environment suitable for use with the present application and is not to be considered as limiting the scope of use of the application.
  • Computer system 900 is also not to be construed as requiring a combination of one or more components in an exemplary computer system 900 that is dependent on or illustrated.
  • 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.

Abstract

本申请提供了一种即时通话方法，所述方法由主叫侧执行并且所述主叫侧为网络电话终端，所述方法包括：向信令服务器发起多方语音通话请求，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；根据所述信令服务器响应所述多方语音通话请求返回的远端访问地址建立数据通道；待指定的终端根据所述远端访问地址建立数据通道之后进行所述多方语音通话；所述终端的数据通道建立中，所述电话终端通过网关设备建立所述远端访问地址对应的数据通道。此外，还提供了一种与该方法匹配的即时通话装置和系统。上述即时通话方法、装置和系统能够实现网络电话终端和电话终端的互通，并且实现由电话终端和网络电话终端参与的多方语音通话。

Description

即时通话方法、装置和系统
本申请要求于2016年1月29日提交中国专利局、申请号为201610066519.9、发明名称为“即时通话方法、装置和系统”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
本申请涉及移动互联网应用技术领域，特别涉及一种即时通话方法、装置和系统。
背景技术
随着网络电话应用的日益广泛使用，用户之间除了在电话终端的配合下通过电话网络进行通话之外，还可通过能够进行互联网络访问的终端设备中运行网络电话应用来实现终端设备之间双方，甚至于多方的语音通话。
由此可知，现有的通话可实现于电话终端之间或者网络电话终端之间，但是，并无法在电话终端和网络电话终端之间实现互通。
具体的，由于电话网络和互联网络是相互独立的，例如，电话网络由运营商经营，互联网络则是由互联网服务商提供服务，因此，电话网络和互联网络互不相干，并无法进行电话终端和网络电话终端之间的互通，更无法实现由电话终端和网络电话终端参与的多方语音通话。
发明内容
本申请的一个目的在于解决现有技术中网络电话终端和电话终端无法互通，并且无法实现由电话终端和网络电话终端参与的多方语音通话的缺陷。
本申请的另一个目的在于提供一种即时通话方法、装置和系统，所述即时通话方法、装置和系统能够实现网络电话终端和电话终端的互通，并且实现由电话终端和网络电话终端参与的多方语音通话。
为解决上述技术问题，本申请采用如下技术方案：
一种即时通话方法，所述方法由主叫侧执行并且所述主叫侧为网络电话 终端，所述方法包括：
向信令服务器发起多方语音通话请求，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；
根据所述信令服务器响应所述多方语音通话请求返回的远端访问地址建立数据通道；以及
待指定的终端根据所述远端访问地址建立数据通道之后进行所述多方语音通话，其中
所述终端的数据通道建立中，所述电话终端通过网关设备建立所述远端访问地址对应的数据通道。
一种即时通话方法，所述方法由被叫侧的网络电话终端执行，所述方法包括：
对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；
根据所述远端访问地址建立所述网络电话终端的数据通道；
待所述多方语音通话请求指定的终端根据所述远端访问地址建立数据通道之后进入所述多方语音通话。
一种即时通话方法，所述方法由信令服务器执行，所述方法包括：
接收主叫侧发起的多方语音通话请求；
响应于所述多方语音通话请求，向主叫侧返回远端访问地址，并向所述多方语音通话请求中指定的终端发起通话建立的信令交互，所述指定的终端包括电话终端；
通过所述信令交互向所述指定的终端中包含的网络电话终端，和/或所述电话终端对应的网关设备返回远端访问地址，以建立多方语音通话。
一种即时通话装置，所述装置运行于主叫侧，并且所述主叫侧为网络电 话终端，所述装置包括：
请求发起模块，用于向信令服务器发起多方语音通话请求，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；
主叫侧通道建立模块，用于根据所述信令服务器响应所述多方语音通话请求返回的远端访问地址建立数据通道；
通话执行模块，用于待指定的终端根据所述远端访问地址建立数据通道之后进行所述多方语音通话；
所述终端的数据通道建立中，所述电话终端通过网关设备建立所述远端访问地址对应的数据通道。
一种即时通话装置，所述装置运行于被叫侧的网络电话终端中，所述装置包括：
请求应答模块，用于对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；
被叫侧通道建立模块，用于根据所述远端访问地址建立所述网络电话终端的数据通道；
待所述多方语音通话请求指定的终端根据所述远端访问地址建立数据通道之后进入所述多方语音通话。
一种即时通话系统，所述系统包括运行于信令服务器中的请求接收模块、通话请求响应模块和通话建立模块，其中：
所述请求接收模块用于接收主叫侧发起的多方语音通话请求；
所述通话请求响应模块用于响应于所述多方语音通话请求，向主叫侧返回远端访问地址，并向所述多方语音通话请求中指定的终端发起通话建立的信令交互，所述指定的终端包括电话终端；
所述通话建立模块用于通过所述信令交互向所述指定的终端中包含的网 络电话终端和/或所述电话终端对应的网关设备返回远端访问地址，以建立多方语音通话。
一种即时通话系统，包括参与多方语音通话的终端、信令服务器、数据服务器和网关设备，所述终端包括网络电话终端和电话终端，其中：
任一网络电话终端向所述信令服务器发起的多方语音通话请求；
向所述发起多方语音通话请求的网络电话终端返回所述数据服务器的远端访问地址，并向所述多方语音通话的其它终端发起通话建立的信令交互，通过所述信令交互向其它网络电话终端和/或电话终端对应的网关设备返回远端访问地址；
所述网络电话终端和/或所述电话终端对应的网关设备通过所述远端访问地址与所述数据服务器建立数据通道，进行多方语音通话
附图说明
图1是一个实施例中即时通话方法的流程图；
图2是另一个实施例中即时通话方法的流程图；
图3是一个实施例中多方语音通话中网络电话终端接收数据通道传输的数据流，并进行自适应混音处理后播放的方法流程图；
图4是图3中对数据流进行自适应判断，输出数据流未进行混音的自适应判断结果的方法流程图；
图5是一个实施例中由被叫侧的网络电话终端执行的即时通话方法的流程图；
图6是图5中对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址的方法流程图；
图7是一个实施例中由信令服务器执行的即时通话方法的流程图；
图8是图7中通过信令交互向指定的终端和/或电话终端对应的网关设备返回远端访问地址，以建立多方语音通话的方法流程图；
图9是另一个实施例中由远端访问地址对应的服务器执行的即时通话方法的流程图；
图10是图9中控制主叫侧和指定的终端之间通过数据通道进行的多方语音通话的方法流程图；
图11是图10中对数据流进行转发处理，以使数据流通过数据通道传输至相应的终端的方法流程图；
图12是一个实施例中多方语音通话的实现拓扑图；
图13是图12中多方语音通话呼叫过程的时序图；
图14是一个实施例中运行于主叫侧的网络电话终端的即时通话装置的结构示意图；
图15是一个实施例中运行于被叫侧的网络电话终端的即时通话装置的结构示意图；
图16是一个实施例中通话接收模块的结构示意图；
图17是一个实施例中运行于被叫侧的网络电话终端的即时通话装置的结构示意图；
图18是图17中请求应答模块的结构示意图；
图19是一个实施例中即时通话系统中信令服务器的结构示意图；
图20是图19中通话建立模块的结构示意图；
图21是一个实施例中远端访问地址对应的服务器的结构示意图；
图22是图21中通话控制模块的结构示意图；
图23是图22中转发处理单元的结构示意图；
图24是图23中自适应混音转发子单元的结构示意图；
图25是本申请的一个实施例中实现即时通话方法的计算机系统的结构示意图。
具体实施方式
为使得本申请的发明目的、特征、优点能够更加的明显和易懂，下面将结合本申请实施例中的附图，对本申请实施例中的技术方案进行清楚、完整地描述，显然，下面所描述的实施例仅仅是本申请一部分实施例，而非全部实施例。基于本申请中的实施例，本领域的技术人员所获得的所有其他实施例，都属于本申请保护的范围。
如前所述的，电话终端和网络电话终端之间所进行的通话存在着绝对隔 离，相互之间无论是从网络上看还是从技术实现上看，均无法互通。
一方面，在网络上，如前所述的，电话网络和互联网络相互独立，各不相干，因此依赖于电话网络的电话终端和依赖于互联网络的网络电话终端将无法实现网络上的互联。
另一方面，在技术实现上，电话网络中，电话终端的通话包括了传输路径、协议、数据实现以及通信接口都是遵循一定的标准的，因此，其电话网络终端具有很好的互联互通性；而网络电话终端则是在其所运行的网络电话应用中通过定制的各种协议完成通话建立和数据实现的，因此网络电话终端能够通过运行的网络电话应用进行通话。
但是，由于协议以及数据实现等技术实现上的各不相同，在技术实现上是难以实现网络电话终端和电话终端的互通的，更无法进行网络电话终端和电话终端参与的多方语音通话。
基于此，将提供一种即时通话方法，以实现网络电话终端和电话终端之间的互通，进而实现网络电话终端和电话终端参与的多方语音通话。
在一个实施例中，由主叫侧执行并且主叫侧为网络电话终端，该即时通话方法如图1所示，包括步骤S110-S150.
步骤S110，向信令服务器发起多方语音通话请求，多方语音通话请求中指定参与多方语音通话的终端包括电话终端。
任意一个网络电话终端都可作为主叫侧，在指定参与多方语音通话的终端之后，即可向信令服务器发起多方语音通话请求。其中，指定参与多方语音通话的终端作为被叫侧，包括任意一个或者任意组合的电话终端，除此之外，还可包括一个或者多个网络电话终端。多方语音通话是指网络电话终端和电话终端这两种类型的终端组合在一起所实现的通话过程，在此通话过程中，网络电话终端通过自身运行的网络电话应用与电话终端进行通话。
作为主叫侧的网络电话终端中，用户在运行的网络电话应用中通过指定若干个号码标识来指定若干个终端作为被叫侧来参与即将发起的多方语音通话。网络电话应用中号码标识的指定过程可以是拨号过程。
信令服务器作为与终端相配合的远端服务器，用于实现信令交互，以控制多方语音通话的建立。
而向信令服务器发起的多方语音通话请求则包括了主叫侧网络电话终端 的终端系统类型、网络类型信息、硬件信息、业务标识、身份标识信息和号码标识，此号码标识包括主叫号码标识和被叫号码标识。
根据多方语音通话请求，可得到当前发起多方语音通话的业务，进而使得网络电话终端中多个应用均可支持多方语音通话功能，将多方语音通话应用于各业务场景中。
步骤S130，根据信令服务器响应多方话音通话请求返回的远端访问地址建立数据通道。
信令服务器在接收多方语音通话请求之后，将相应地向网络电话终端返回远端访问地址。其中，所返回的远端访问地址是一服务器所在的地址，该服务器将用于对多方语音通话中数据流的传输进行控制。
需要说明的是，该服务器可以与信令服务器为同一服务器，也可以是区别于信令服务器的其它服务器，具体的服务器和信令服务器布局情况可根据实际需要进行灵活配置。
发起多方语音通话请求的网络电话终端直接根据远端访问地址与服务器建立数据通道，该数据通道用于实现与网络电话终端相关的数据流的传输。具体可包括网络电话终端向服务器发送输入而产生的数据流，或者服务器向网络电话终端返回来自其终端的数据流。
由此可知，信令服务器的作用在于为各终端接入多方语音通话，而远端访问地址对应的服务器的作用则在于为多方语音通话中各终端之间产生的数据流提供中转。
步骤S150，待指定的终端根据远端访问地址建立数据通道之后进行多方语音通话。
终端的数据通道建立中，电话终端通过网关设备建立远端访问地址对应的数据通道。
主叫侧的网络电话终端在完成了数据通道的建立之后，还将进行预设时间的等待，以等待指定的终端响应多方语音通话请求而建立数据通道之后，方可进入多方语音通道。
在此需要特别说明的是，在主叫侧的网络电话终端进行预设时间的等待中，可以是指定的某一个或者几个终端完成了数据通道的建立，也可以是指定的所有终端都完成了数据通道的建立，无论何种情况，都可以进行发起的 多方语音通话。
换而言之，在主叫侧的网络电话终端建立数据通道的前提下，任意一个或者任意组合的终端只要完成数据通道的建立，都可参与到主叫侧的网络电话终端发起的多方语音通话中。
被叫侧，对于网络电话终端，其可直接根据远端访问地址与对应的服务器建立数据通道；对于电话终端，由于电话网络与互联网络是相互独立的，因此，需要网关设备来作为电话网络和互联网络之间的边界设备，网关设备将根据远端访问地址与对应的服务器建立电话终端的数据通道，以使电话终端得以通过网关设备与服务器建立连接。
电话终端包括固定电话和移动电话。与电话终端相配合的，网关设备为Voip网关，其一侧连接电话网络，另一侧则连接到互联网络中，以实现两个网络之间协议和数据的转换，进而实现两个网络的互通。
因此，在网关设备的作用下，将使得电话网络和互联网络能够网际互联，从而实现网络电话终端和电话终端的通话，甚至于在网络电话终端的辅助下，使得电话终端也能够参与到网络电话终端所进行的多方语音通话中，解除了通话的界限和限制。
通过如上所述的过程，实现了网络电话终端和电话终端之间的语音通话，甚至于包括了电话终端在内的多个终端参与的多方语音通话，从而在电话终端中实现了其本不支持的多方语音通话功能。
随着互联网络中移动网络接入越来越方便，带宽成本的降低和传输质量的保障，使得网络电话终端和电话终端之间能够随意实现多方语音通话，进而使得语音通话的实现不再仅限于电话网络，极大地扩展了通话的实现。
在一个实施例中，如上所述的方法如图2所示，还包括步骤S210和S230。
步骤S210，指定参与多方语音通话的各网络电话终端对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址。
可以理解的，指定参与多方语音通话的各网络电话终端为被叫侧，相应的，转发的多方语音通话请求对于被叫侧的网络电话终端而言，将作为呼叫请求，以在被叫侧的网络电话终端中进行振铃通知，并等待应答。
具体的，网络电话终端在接收到多方语音通话请求之后将进行振铃，以对发起的多方语音通话呼叫进行通知。
此时，对振铃所进行的响应即为多方语音通话请求的应答，其包括接听、拒绝和超时挂断，在检测到接听操作时网络电话终端时进行应答以得到信令服务器返回的远端访问地址。
步骤S230，根据远端访问地址建立网络电话终端的数据通道。
指定参与多方语音通话的各网络电话终端在得到远端访问地址之后，将按照远端访问地址建立各自的数据通道，进而由此实现通话的建立。
通过如上过程，将使得主叫侧的网络设备发起多方语音通话时，被叫侧的任意网络电话终端通过应答即可参与到主叫侧与电话终端的语音通话中，从而使得语音通话具备较强的灵活性。
在一个实施例中，如上所述的方法还包括：多方语音通话中网络电话终端接收数据通道传输的数据流，并进行自适应混音处理后播放。
也就是说，网络电话终端和电话终端参与的多方语音通话中，任意网络电话终端都将进行数据流的接收，其中，所接收到的数据流可以是一路数据流，也可以是多路数据流，因此，需要进行自适应混音处理。
一方面，由于涉及多方语音通话，因此，对于一个通过自身建立的数据通道接收数据流的网络电话终端而言，其接收的数据流可能是一路数据流，例如，当前只有一终端进行语音输入，但也有可能是多路数据流，即参与语音通话的其它终端中某几个终端或者所有终端都进行语音输入。
另一方面，多方语音通话中涉及的多路数据流混音，其是在远端或者网络电话终端本地进行自适应地动态切换的，因此，如果远端已进行了混音，则网络电话终端接收到的数据流则必然是一路数据流；如果远端未进行混音，则网络电话终端接收到的数据流为多路数据流，并且需要对多路数据流进行混音。
多方语音通话中，无论主叫侧还是被叫侧，任意网络电话终端在完成了数据流的自适应混音处理之后方可进行播放，以完成多方语音通话中的语音输出。
其中，自适应混音处理包括了数据流是否混音的自适应判断过程以及基于自适应判断结果的执行过程。
通过此过程来动态自适应支持远端的服务器混音和终端本地混音两种方式，进而减轻了多方语音通话中服务器的负载，也降低了网络电话终端进行 数据流接收的延迟，进而使得当前通过电话网络和互联网络所实现的多方语音通话能够具备与电话网络中通话相当的时效性和顺畅性。
进一步的，在本实施例中，如图3所示，上述多方语音通话中网络电话终端接收数据通道传输的数据流，并进行自适应混音处理后播放的步骤包括步骤S310-350。
步骤S310，多方语音通话中，各网络电话终端通过数据通道接收数据流，数据流来自其它终端。
如前所述的，多方语音通话是多个终端之间的通话，此多个终端包括了网络电话终端和电话终端，并且都建立了各自的数据通道。
各终端完成了数据通道建立进入多方语音通话时，网络电话终端都会通过其自身建立的数据通道进行数据流的接收，以获知当前其它终端所进行的语音输入。
步骤S330，对数据流进行自适应判断，输出数据流未进行混音的自适应判断结果。
身份标识信息标识数据流所来自的终端，除此之外，进行了服务器混音的数据流还通过身份标识信息来标识其所对应的目标终端，目标终端为接收该数据流的终端。
换而言之，未进行服务器混音的数据流中，任意一路数据流都附带了标记所来自的终端的身份标识信息；进行了服务器混音所得到的一路数据流中，除了附带标识所来自终端的身份标识信息之外，还附带了对应的目标终端的身份标识信息。
因此，对于网络电话终端接收的数据流而言，将以附带的身份标识信息为依据来进行自适应判断。
步骤S350，根据自适应判断结果进行数据流的混音处理，并播放混音处理所得到的数据流。
如果输出数据流未进行混音的自适应判断结果，则会在网络电话终端进行终端本地混音，以将接收的数据流叠加为一路数据流之后，再进行播放，完成多方语音通话中的语音输出。
如上所述的自适应判断和混音处理过程，为服务器混音和终端本地混音的动态自适应切换提供了具体实现，进而增强了多方语音通话中数据实现的 灵活性。
进一步的，在本实施例中，步骤S330如图4所示，包括步骤S331-335。
步骤S331，判断数据流是否附带自身的身份标识信息，若为否，则进入步骤S333，若为是，则进入步骤S335。
进行了数据流接收的网络电话终端中，对接收的任意一路数据流，都判断其是否附带本网络电话终端自身的身份标识信息，如果是，则说明该数据流是进行了服务器混音后传输的，由此将相应输出数据流已进行混音的自适应判断结果，反之则输出数据流未进行混音的自适应判断结果。
步骤S333，输出数据流未进行混音的自适应判断结果。
步骤S335，输出数据流已进行混音的自适应判断结果，该自适应判断结果触发进行数据流的播放。
如前所述的，多方语音通话中涉及了身份标识信息和号码标识，其中，身份标识信息用于以在数据流中标识来自的终端以及服务器混音所针对的目标终端；号码标识则用于标记数据流传输所对应的源终端和目标终端。所说的目标终端对于相同数据流而言，号码标识和身份标识信息所涉及的目标终端将为同一终端。
在另一个实施例中，如上所述的方法还包括：多方语音通话中任意网络电话终端编码输入通话数据得到自身输入的数据流，并通过建立的数据通道传输。
多方语音通话中，任意网络电话终端都会进行语音输入，此时，将对语音输入所得到的通话数据进行编码以形成自身输入的数据流，该数据流需要通过自身建立的数据通道传输至远端访问地址所对应的服务器中，进而在服务器的控制下将数据流送达参与多方语音通话的多个终端。
也就是说，任意网络电话终端都建立了自身所对应的数据通道，并且通过此数据通道实现数据流的传输，由此实现多方语音通话中各网络电话终端的数据流产生以及传输。
通过如上所述的过程，提供了一种能够发起多方语音通话并且在多方语音通话中实现数据流的传输和处理的网络电话终端，除此之外，还提供了一种能够参与多方语音通话的网络电话终端，电话终端则在网关设备的配合下接入多方语音通话，由此便构成了互联网络和电话网络所支持的多方语音通 话，进而使得网络电话终端和电话终端能够自由通话。
另外，对于电话终端而言，其是借助网络电话终端来实现多方语音通话的支持，也就是说，通过网络电话终端中多方语音通话请求的发起，使得电话终端也能够在其原本不支持多方语音通话的基础上参与到网络电话终端发起的多方语音通话中，极大地扩展了电话终端的功能。
在一个实施例中，还相应地提出了一种即时通话方法，该方法由被叫侧的网络电话终端执行，如图5所示，包括步骤S410-450。
步骤S410，对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址，多方语音通话请求中指定参与多方语音通话的终端包括电话终端。
步骤S430，根据远端访问地址建立网络电话终端的数据通道。
步骤S450，待多方语音通话请求指定的终端根据远端访问地址建立数据通道之后进入多方语音通话。
通过上述步骤，实现了网络电话终端中应答其它网络电话终端的多方语音通话请求而进入多方语音通话的过程，由此将使得作为被叫侧的任意网络电话终端都将通过如上所述的步骤进行应答，以进入多方语音通话。
进一步的，在本实施例中，如图6所示，该步骤S410包括S411-S415。
步骤S411，接收信令服务器为主叫侧转发的多方语音通话请求。
如前所述的，信令服务器用以控制多方语音通话的建立，其在接收到多方语音通话请求之后，将按照多方语音通话请求中携带的被叫号码标识进行转发。
步骤S413，根据检测的接听操作生成接听应答信令，并将接听应答信令传送至信令服务器。
接收多方语音通话请求的网络电话终端将对此进行振铃通知，此时，如果检测到拒绝操作，则中断当前所进行的振铃通知；如果进行的振铃通知等待超时则触发挂断操作，也中断当前所进行的振铃通知，并且拒绝操作和挂断操作的触发都将停止当前所进行的多方语音通话建立过程。
如果在当前进行的振铃通知中检测到接听操作，则生成接听应答信令，并传送至信令服务器中。
步骤S415，接收信令服务器根据接听应答信令发送的远端访问地址。
信令服务器在接收到网络电话终端的接听应答信令之后便向其下发远端访问地址，通过远端访问地址向网络电话终端所进行的多方语音通话提供能够进行数据流中转的节点，进而通过远端访问地址连接多方语音通话中的各终端，由此方可顺利进行多方语音通话。
在一个实施例中，还相应地提供了一种即时通话方法，该方法由信令服务器执行，如图7所示，包括步骤S510-S550。
步骤S510，接收主叫侧发起的多方语音通话请求。
相对作为主叫侧的网络电话终端，信令服务器在接收到任意网络电话终端发起的多方语音通话请求，从而根据多方语音通话请求中携带的标识号码即可获知发起多方语音通话请求的网络电话终端以及指定参与多方语音通话的网络电话终端和电话终端。
步骤S530，响应于多方语音通话请求，向主叫侧返回远端访问地址，并向多方语音通话请求中指定的终端发起通话建立的信令交互，指定的终端包括电话终端。
信令服务器在接收到多方语音通话请求之后，既应当对主叫侧进行响应，也应当对相关的终端，即通过多方语音通话请求向指定的终端发起呼叫。
具体的，一方面，信令服务器按照多方语音通话请求中携带的主叫号码标识向主叫侧返回远端访问地址，以触发主叫侧网络电话终端的数据通道建立。
另一方面，信令服务器按照多方语音通话请求中携带的被叫号码标识向相应的终端发起通话建立的信令交互，以触发信令服务器和终端之间的应答以及数据通道建立过程。
需要进一步说明的是，对于信令服务器和指定的终端之间的应答以及数据通道建立过程，如果该终端是电话终端，则是通过网关设备来实现的。
步骤S550，通过信令交互向指定的终端和/或电话终端对应的网关设备返回远端访问地址，以建立多方语音通话。
信令服务器一方面向主叫侧的网络电话终端发送远端访问地址，另一方面，指定的终端包含了网络电话终端和/或电话终端，从而也向网络电话终端和/或电话终端对应的网关设备发送远端访问地址，以触发被叫侧的网络电话 终端和/或电话终端的多方语音通话建立过程。
通过如上所述的信令服务器实现了各终端之间多方语音通话的建立，在信令服务器的作用下为网络电话终端和电话终端建立多方语音通话，进而实现各终端之间的通话，此通话能够将电话终端加入由网络电话终端所发起的多方语音通话中，为电话终端进行多方语音通话提供了一种可行的信令服务器，从而通过信令服务器使得网络电话终端和电话终端之间具备很好的互联互通性，网络电话终端和电话终端之间能够自由通话。
在一个实施例中，信令服务器向被叫侧的各终端发起多方语音通话建立的信令交互之间，还将识别本次发起的多方语音通话请求中指定的各终端所对应类型，以得到相应的终端识别结果。
终端识别结果将指示被叫侧的相应终端是网络电话终端还是电话终端，因此，将根据终端识别结果发起信令交互。
例如，根据终端识别结果，可对网络电话终端直接进行多方语音通话请求的转发，对电话终端则需要转换为电话呼叫请求之后再进行发送。
在优选的实施例中，终端的识别将是根据被叫标识号码进行的。具体的，判断被叫标识号码是否为已注册的标识号码，若为是，则得到终端为网络电话终端的终端识别结果，若为否，则得到终端为电话终端的终端识别结果。
也就是说，对于任意网络电话终端，其在登录之后，就已经在信令服务器中注册其所对应的标识号码，以在信令服务器中维护其所处的在线状态，进而能够随时进行通话。
进一步的，在本实施例中，如图8所示，步骤S550包括步骤S551-S555。
步骤S551，为电话终端转换多方语音通话请求得到电话呼叫请求，并发送到对应的网关设备，电话呼叫请求通过网关设备送达电话终端。
多方语音通话请求中指定的终端包括电话终端，由于互联网络和电话终端中涉及的协议和数据实现各不相同，因此，需要对多方语音通话请求进行转换，以转换成网关设备和电话终端能够识别的电话呼叫请求。
信令服务器在为电话终端进行多方语音通话请求的转发得到电话呼叫请求之后，向网关设备发送得到的电话呼叫请求，以使得电话呼叫请求能够通过网关设备送达电话终端。
步骤S553，接收网关设备转发的接听应答信令，接听应答信令是电话终 端响应电话呼叫请求返回至网关设备的。
网关设备在接收到信令服务器发关的电话呼叫请求之后，将通过电话呼叫请求对电话终端进行振铃通知，以等待电话终端进行应答。
进行振铃通知的电话终端中，如果检测到接听操作，则相应生成接听应答信令，发送至网关设备，以通过网关设备代为向信令服务器上报电话终端的应答。
步骤S555，响应于接听应答信令，向网关设备返回远端访问地址，以控制网关设备通过远端访问地址建立多方语音通话。
信令服务器在获知电话终端中触发的接听操作时会通过向网关设备返回远端访问地址来进行多方语音通话的建立。
具体的，通过信令服务器所进行的远端访问地址发送，网络设备为电话终端建立数据通道，以完成电话终端这一侧的多方语音通话建立。
网络电话终端是直接与远端访问地址所对应的服务器建立数据通道，区别于网络电话终端，电话终端则是由网关设备与远端访问地址所对应的服务器建立数据通道。
相对于电话终端而言，通过信令服务器、网关设备以及电话终端三者之间的信令交互来实现信令服务器和网关设备之间数据通道的建立，进而得以将电话终端引入多方语音通话中，信令服务器所成功实现的电话终端引入极大增强了通话性能。
在一个实施例中，如图9所示，该方法还包括步骤S610和S630。
步骤S610，远端访问地址对应的服务器与主叫侧、指定的终端分别建立数据通道。
在信令服务器通过发送远端访问地址来触发各终端中数据通道的建立之后，将在远端访问地址对应的数据中与各终端分别建立数据通道。需要说明的是，各终端包括作为主叫侧的网络电话终端，还包括作为被叫侧的网络电话终端和/或电话终端。
步骤S630，控制主叫侧和指定的终端之间通过数据通道进行多方语音通话。
通过各终端建立的数据通道，远端访问地址对应的服务器会接收到数据流，此数据流来自于多方语音通话的任意一终端，此时，将对数据流进行相 关的处理以通过各终端的数据通道使得多方语音通话的各终端能够得到来自于其它终端的数据流，由此即可进行各终端之间的多方语音通话。
远端访问地址对应的服务器可以是数据服务器，其用于进行多方语音通话中的数据实现，例如传输控制等，其可以与信令服务器搭载于同一服务器设备中，也可以分别搭载于不同的服务器设备中，根据实际运营的需要配置服务器的拓扑。
进一步的，如图10所示的，该步骤S630包括步骤S631和S633。
步骤S631，接收多方语音通话中任意网络电话终端发送的数据流或电话终端通过网关设备发送的数据流。
在数据通道的作用下，来自任意网络电话终端或者电话终端的数据流都被传输至远端访问地址对应的服务器中。
步骤S633，对数据流进行转发处理，以使数据流通过数据通道传输至相应的终端。
远端访问地址对应的服务器中，对于一多方语音通话，所接收的数据流为一路或者多路，因此，需要进行转发处理。其中，该转发处理可以是直接转发，也可以是混音和转发两个操作。
例如，对于一路数据流，直接对此进行转发即可；对于多路数据流，则需要进行混音后方可转发。
在一个实施例中，接收的数据流为多路数据流，则步骤S633如图11所示，包括步骤S6331和S6333。
步骤S6331，确定多路数据流的接收端，接收端包括网络电话终端和/或电话终端。
远端访问地址对应的服务器中，对于一多方语音通话，其当前将接收到多路数据流，此时，按照多路数据流所对应的多个接收端分别进行处理。
具体的，根据多路数据流中附带的身份标识信息，此身份标识信息所对应的终端之外的其它终端即为接收端。
步骤S6333，针对每一接收端，接收端为网络电话终端且支持混音处理时转发多路数据流中接收端对应的数据流，接收端为电话终端或者接收端为网络电话终端且不支持混音处理时将转发的数据流进行混音处理后转发。
转发的数据流是任意一路数据流或者任意组合的数据流。
所接收的多路数据流中，每一路数据流均有对应的接收端，因此，在所确定得到的多个接收端中，每一接收端都有对应的一路或者组合的几路数据流，因此将分别为每一接收端处理其所对应的数据流。
具体的，网络电话终端或者网关在完成数据通道建立进入多方语音通话时，还将就其自身是否支持混音处理向远端访问地址对应的服务器进行通知。在此基础上，如果一接收端为网络电话终端且支持混音处理，则直接进行对应数据流的转发；如果一接收端为电话终端或者接收端为网络电话终端且不支持混音处理，则需要对数据流进行混音处理，以使得多路数据流能够传输至电话终端和不支持混音处理的网络电话终端，进而保证仅能够接收一路数据流的电话终端能够进行多方语音通话，也进一步兼容了各种类型的网络电话终端，提高了多方语音通话的应用范围。
在一个实施例中，对电话终端所进行的混音处理还包括电话格式的编码，转发的数据流是电话终端支持的码流，该方法还包括：
网关设备接收电话终端支持的码流，并将码流送达相应的电话终端。
远端访问地址对应的服务器除了对即将传输至电话终端的数据流进行混音之外，还将进行编码，以使得最终转发至网关设备，进而由网关设备转发至相应电话终端的数据流是网络设备和电话终端所能够识别的，进而符合电话网络中的协议和数据实现。
在一个实施例中，步骤S6333中，接收端为电话终端或者接收端为网络电话终端且不支持混音处理时将待转发的数据流进行混音处理后转发的过程为：
接收端电话终端或者接收端为网络电话终端且不支持混音处理时，叠加待转发的数据流以输出一路数据流，在输出的一路数据流中附带接收端的身份标识信息。
向接收端转发附带接收端的身份标识信息的一路数据流。
在此过程中，数据流的叠加可通过多路数字信号时域的线性叠加实现。在完成叠加所得到的一路数据流中，附带接收端的身份标识信息，以使得接收端能够通过此身份标识信息来简易快速地判断是否已进行服务器混音。
通过如上所述的过程，将为多方语音通话的实现提供了服务器以及网关设备的系统架构及实现，从而能够实现包含网络电话终端和电话终端在内的 多个终端之间的多方语音通话。
下面结合一个具体的实施例来详细阐述如上所述的多方语音通话。在本实施例中，该多方语音通话包括了n个终端，其中，如图12所示，n是不小于4的整数，以这一多方语音通话为例进行阐述。
n个终端将至少包括网络电话终端1、网络电话终端2、电话终端1和电话终端2。多方语音通话包括了多方语音通话呼叫过程和多方语音通话建立后的数据处理过程。
首先，一网络电话终端770作为主叫侧来向信令服务器710发起多方语音通话请求，其它网络电话终端770和电话终端790作为被叫侧，在信令服务器710的控制下进行多方语音通话请求的应答，即执行如图13所示的时序过程，进而使得各终端均能够由信令服务器710得到远端访问地址，进而建立各终端的数据通道，完成多方语音通话呼叫过程。
远端访问地址为数据服务器730的地址。
任意网络电话终端770中，编码输入的通话数据得到自身输入的数据流，并通过数据通道转发到数据服务器730。
任意电话终端790中输入的语音数据被发送到Voip网关750，由Voip网关750转发到数据服务器730，数据服务器730立即对此进行转发，以得到网络电话终端770能够解析的数据流。
基于此，在数据服务器730的一次接收中，将接收到多路数据流。数据服务器730确定多路数据流将发送至哪些终端中，即进行接收端的确定；然后在对每一接收端进行直接转发或者混音处理后转发。
如果接收端为电话终端790，数据服务器30进行混音处理后进行数据流的转发，而此混音处理还包括了按照电话要求的格式编码的过程，以使得电话终端790能够识别的数据流。
通过如上所述的过程，既实现了电话终端790的多方语音通话，又在多方语音通话中支持混音的自适应动态切换，极大地增强了通话性能。
如上所述的网络电话终端770可以是各种移动终端和固定终端，例如，智能手机、平板电话、台式电脑等，任意能够访问互联网络且运行网络电话应用的终端都能够作为网络电话终端770。
在一个实施例中，还相应地提供了一种即时通话装置，该装置运行于主叫侧，并且主叫侧为网络电话终端，如图14所示，该装置包括请求发起模块110、主叫侧通道建立模块130和通话执行模块150，其中：
请求发起模块110，用于向信令服务器发起多方语音通话请求，多方语音通话请求中指定参与多方语音通话的终端包括电话终端。
主叫侧通道建立模块130，用于根据信令服务器响应多方语音通话请求返回的远端访问地址建立数据通道。
通话执行模块150，用于待指定的终端根据远端访问地址建立数据通道之后进行多方语音通话。
终端的数据通道建立中，电话终端通过网关设备建立远端访问地址对应的数据通道。
在一个实施例中，如上所述的装置如图15所示，还包括请求应答模块210和被叫侧通道建立模块230，请求应答模块210和被叫侧通道建立模块230运行于指定参与方语音通话的各网络电话终端中，其中：
请求应答模块210，用于对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址。
被叫侧通道建立模块230，用于根据远端访问地址建立网络终端的数据通道。
在另一个实施例中，如上的装置还包括了运行于多方语音通话中网络电话终端的通话接收模块，该通话接收模块用于接收数据通道传输的数据流，并进行自适应混音处理后播放。
进一步的，在本实施例中，如图16所示，通话接收模块300包括接收执行单元310、自适应判断单元330和本地混音单元350，其中：
接收执行单元310，用于通过数据通道接收数据流，数据流来自其它终端。
自适应判断单元330，用于对数据流进行自适应判断，输出数据流未进行混音的自适应判断结果。
本地混音单元350，用于根据自适应判断结果进行数据流的混音处理，并播放混音处理所得到的数据流。
进一步的，自适应判断单元330用于判断数据流是否附带自身的身份标 识信息，若为否，则输出数据流未进行混音的自适应判断结果，若为是，则输出数据流已进行混音的自适应判断结果，自适应判断结果触发进行数据流的播放。
在一个实施例中，如上所述的装置还包括数据流输入模块，该数据流输入模块运行于多方语音通话中的任意网络电话终端，用于编码输入的通话数据得到自身输入的数据流，并通过建立的数据通道传输。
在一个实施例中，还相应地提供了一种即时通话装置，该装置运行于被叫侧的网络电话终端中，该装置如图17所示，包括请求应答模块410和被叫侧通话建立模块430。
请求应答模块410，用于对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址，多方语音通话请求中指定参与多方语音通话的终端包括电话终端。
被叫侧通话建立模块430，用于根据远端访问地址建立网络电话终端的数据通道。
待多方语音通话请求指定的终端根据远端访问地址建立数据通道之后进入多方语音通话。
进一步的，如图18所示，请求应答模块410包括转发请求接收单元411、信令生成单元413和地址接收单元415，其中：
转发请求接收单元411，用于接收信令服务器为主叫侧转发的多方语音通话请求。
信令生成单元413，用于根据检测的接听操作生成接听应答信令，并将接收应答信令传送至信令服务器。
地址接收单元415，用于接收信令服务器根据接听应答信令发送的远端访问地址。
在一个实施例中，还相应地提供了一种即时通话系统，如图19所示，该系统包括运行于信令服务器中的请求接收模块510、通话请求响应模块530和通话建立模块550，其中：
请求接收模块510，用于接收主叫侧发起的多方语音通话请求。
通话请求响应模块530，用于响应于多方语音通话请求，向主叫侧返回远端访问地址，并向多方语音通话请求中指定的终端发起通话建立的信令交互，指定的终端包括电话终端。
通话建立模块550，用于通过信令交互向指定的终端中包含的网络电话终端和/或电话终端对应的网关设备返回远端访问地址，以建立多方语音通话。
进一步的，在本实施例中，通话建立模块550如图20所示，包括请求转换单元551、应答信令接收单元553和信令响应单元555，其中：
请求转换单元551，用于为电话终端转换多方语音通话请求得到电话呼叫请求，并发送到对应的网关设备，电话呼叫请求通过网关设备送达电话终端。
应答信令接收单元553，用于接收网关设备转发的接听应答信令，接听应答信令是电话终端响应电话呼叫请求返回至网关设备的。
信令响应单元555，用于响应于接听应答信令，向网关设备返回远端访问地址，以控制网关设备通过远端访问地址建立多方语音通话。
在一个实施例中，如图21所示，该系统还包括通道建立模块610和通话控制模块630，通道建立模块610和通话控制模块630运行于远端访问地址对应的服务器中。
通道建立模块610，用于与主叫侧、指定的终端分别建立数据通道。
通话控制模块630，用于控制主叫侧和指定的终端之间通过数据通道进行的多方语音通话。
进一步的，在本实施例中，如图22所示，通话控制模块630包括数据流接收单元631和转发处理单元633，其中：
数据流接收单元631，用于接收多方语音通话中任意网络电话终端发送的数据流或电话终端通过网关设备发送的数据流。
转发处理单元633，用于对数据流时行转发处理，以使数据流通过数据通道传输至相应的终端。
进一步的，接收的数据流为多路数据流，转发处理单元633如图23所示，包括接收端确定子单元6331和自适应混音转发子单元6333，其中：
接收端确定子单元6331，用于确定多路数据流的接收端，其中所述接收 端包括网络电话终端和/或电话终端。
自适应混音转发子单元6333，用于当接收端为网络电话终端且支持混音处理时，转发多路数据流中接收端对应的数据流，当接收端为电话终端或者接收端为网络电话终端且不支持混音处理时，将待转发的数据流进行混音处理后转发。
转发的数据流是任意一路数据流或者任意组合的数据流。
进一步的，对电话终端进行的混音处理包括电话格式的编码，转发的数据流是电话终端支持的码流，则该系统还包括网关设备，网关设备用于接收电话终端支持的码流，并将码流送达相应的电话终端。
在一个实施中，如图24所示，自适应混音转发子单元6333包括数据流叠加子单元63331和转发执行子单元63333，其中：
数据流叠加子单元63331，用于当接收端为电话终端或者接收端为网络电话终端且不支持混音处理时，叠加待转发的数据流以输出一路数据流，并在输出的一路数据流中附带接收端的身份标识信息；
转发执行子单元63333，用于向接收端转发附带接收端的身份标识信息的一路数据流。
在一个实施例中，还相应地提供了一种即时通话系统，请结合参阅图12，该系统包括参与多方语音通话的终端、信令服务器710、数据服务器730和网关设备750，所述终端包括网络电话终端770和电话终端790，其中：
任一网络电话终端770向信令服务器710发起的多方语音通话请求。
向发起多方语音通话请求的网络电话终端770返回数据服务器730的远端访问地址，并向多方语音通话的其它终端发起通话建立的信令交互，通过信令交互向其它网络电话终端770和/或电话终端790对应的网关设备750返回远端访问地址。
网络电话终端770和/或所述电话终端90对应的网关设备750通过远端访问地址与数据服务器730建立数据通道，进行多方语音通话。
本实施例提供的一种即时通话方法由计算机程序实现，该即时通话装置和系统则被存储于如图25所述的计算机系统900中，以在该计算机系统900 中运行。
如图25所示，该计算机系统900包括处理器910、存储器920和系统总线922。包括存储器920和处理器910在内的各种系统组件连接到系统总线1022上。处理器910是一个用来通过计算机系统中基本的算术和逻辑运算来执行计算机程序指令的硬件。存储器920是一个用于临时或永久性存储计算程序或数据(例如，程序状态信息)的物理设备，存储了语音通话中系统电话的处理方法的程序指令以及其它数据，处理器910将执行存储器920中的程序指令，并监听系统电话的状态变化，并对此进行响应，以实现语音通话的暂时中断或恢复。处理器910和存储器920可以通过系统总线1022进行数据通信。其中存储器920包括只读存储器(ROM)或闪存(图中都未示出)，以及随机存取存储器(RAM)，RAM通常是指加载了操作系统和应用程序的主存储器。
计算机系统900还包括显示接口930(例如，图形处理单元)、显示设备940(例如，液晶显示器)、音频接口950(例如，声卡)以及音频设备960(例如，扬声器)。显示设备940将是用于进行语音通话界面的显示。
计算机系统900一般包括一个存储设备970。存储设备970可以从多种计算机可读介质中选择，计算机可读介质是指可以通过计算机系统900访问的任何可利用的介质，包括移动的和固定的两种介质。例如，计算机可读介质包括但不限于，闪速存储器(微型SD卡)，CD-ROM，数字通用光盘(DVD)或其它光盘存储、磁带盒、磁带、磁盘存储或其它磁存储设备，或者可用于存储所需信息并可由计算机系统900访问的任何其它介质。
计算机系统900还包括输入装置980和输入接口990(例如，IO控制器)。用户可以通过输入装置980，如键盘、鼠标、显示装置940上的触摸面板设备，输入指令和信息到计算机系统900中。输入装置980通常是通过输入接口990连接到系统总线922上的，但也可以通过其它接口或总线结构相连接，如通用串行总线(USB)。
计算机系统900可在网络环境中与一个或者多个网络设备进行逻辑连接。网络设备可以是个人电脑、服务器、路由器、智能电话、平板电脑或者其它公共网络节点。计算机系统900通过局域网(LAN)接口1000或者移动通信单元1010与网络设备相连接。局域网(LAN)是指在有限区域内， 例如家庭、学校、计算机实验室、或者使用网络媒体的办公楼，互联组成的计算机网络。WiFi和双绞线布线以太网是最常用的构建局域网的两种技术。WiFi是一种能使计算机系统900间交换数据或通过无线电波连接到无线网络的技术。移动通信单元1010能在一个广阔的地理区域内移动的同时通过无线电通信线路接听和拨打电话。除了通话以外，移动通信单元1010也支持在提供移动数据服务的2G，3G或4G蜂窝通信系统中进行互联网访问。
如上面详细描述的，适用于本申请的计算机系统900能执行上述即时通话的指定操作。计算机系统900通过处理器910运行在计算机可读介质中的程序指令的形式来执行这些操作。这些程序指令可以从存储设备970或者通过局域网接口1000从另一设备读入到存储器920中。存储在存储器920中的程序指令使得处理器910执行上述的实现图像处理的方法。此外，通过硬件电路或者硬件电路结合软件指令也能同样实现本申请。因此，实现本申请并不限于任何特定硬件电路和软件的组合。
以上计算机系统900只是一个适用于本申请的计算机环境的示例，不能认为是提出了对本申请的使用范围的任何限制。计算机系统900也不能解释为需要依赖于或具有图示的示例性的计算机系统900中的一个或多个部件的组合。
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。
虽然已参照几个典型实施方式描述了本申请，但应当理解，所用的术语是说明和示例性、而非限制性的术语。由于本申请能够以多种形式具体实施而不脱离发明的精神或实质，所以应当理解，上述实施方式不限于任何前述的细节，而应在随附权利要求所限定的精神和范围内广泛地解释，因此落入权利要求或其等效范围内的全部变化和改型都应为随附权利要求所涵盖。

Claims (31)

  1. 一种即时通话方法，其特征在于，所述方法由主叫侧执行并且所述主叫侧为网络电话终端，所述方法包括：
    向信令服务器发起多方语音通话请求，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；
    根据所述信令服务器响应所述多方语音通话请求返回的远端访问地址建立数据通道；以及
    待指定的终端根据所述远端访问地址建立数据通道之后进行所述多方语音通话，其中所述终端的数据通道建立中，所述电话终端通过网关设备建立所述远端访问地址对应的数据通道。
  2. 根据权利要求1所述的方法，其特征在于，所述方法包括：
    指定参与多方语音通话的各网络电话终端对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址；以及
    根据所述远端访问地址建立所述网络电话终端的数据通道。
  3. 根据权利要求1或2所述的方法，其特征在于，所述方法还包括：
    所述多方语音通话中网络电话终端接收数据通道传输的数据流，并进行自适应混音处理后播放。
  4. 根据权利要求3所述的方法，其特征在于，所述多方语音通话中网络电话终端接收数据通道传输的数据流，并进行自适应混音处理后播放的步骤包括：
    所述多方语音通话中，各网络电话终端通过所述数据通道接收数据流，所述数据流来自其它终端；
    对所述数据流进行自适应判断，输出所述数据流未进行混音的自适应判断结果；以及
    根据所述自适应判断结果进行所述数据流的混音处理，并播放混音处理所得到的数据流。
  5. 根据权利要求4所述的方法，其特征在于，所述对所述数据流进行自适应判断，输出所述数据流未进行混音的自适应判断结果的步骤包括：
    判断所述数据流是否附带自身的身份标识信息，若为否，则输出所述数据流未进行混音的自适应判断结果，若为是，则
    输出所述数据流已进行混音的自适应判断结果，所述自适应判断结果触发进行所述数据流的播放。
  6. 根据权利要求1所述的方法，其特征在于，所述方法还包括：
    所述多方语音通话中任意网络电话终端编码输入通话数据得到自身输入的数据流，并通过建立的所述数据通道传输。
  7. 一种即时通话方法，其特征在于，所述方法由被叫侧的网络电话终端执行，所述方法包括：
    对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；
    根据所述远端访问地址建立所述网络电话终端的数据通道；以及
    待所述多方语音通话请求指定的终端根据所述远端访问地址建立数据通道之后进入所述多方语音通话。
  8. 根据权利要求7所述的方法，其特征在于，所述对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址的步骤包括：
    接收信令服务器为主叫侧转发的多方语音通话请求；
    根据检测的接听操作生成接听应答信令，并将所述接听应答信令传送至所述信令服务器；以及
    接收所述信令服务器根据所述接听应答信令发送的远端访问地址。
  9. 一种即时通话方法，其特征在于，所述方法由信令服务器执行，所述方法包括：
    接收主叫侧发起的多方语音通话请求；
    响应于所述多方语音通话请求，向主叫侧返回远端访问地址，并向所述多方语音通话请求中指定的终端发起通话建立的信令交互，所述指定的终端包括电话终端；以及
    通过所述信令交互向所述指定的终端中包含的网络电话终端，和/或所述电话终端对应的网关设备返回远端访问地址，以建立多方语音通话。
  10. 根据权利要求9所述的方法，其特征在于，所述通过所述信令交互向所述指定的终端中包含的所述电话终端对应的网关设备返回远端访问地址，以建立多方语音通话的步骤包括：
    为所述电话终端转换所述多方语音通话请求得到电话呼叫请求，并发送到对应的网关设备，所述电话呼叫请求通过所述网关设备送达所述电话终端；
    接收网关设备转发的接听应答信令，所述接听应答信令是所述电话终端响应所述电话呼叫请求返回至所述网关设备的；以及
    响应于所述接听应答信令，向所述网关设备返回远端访问地址，以控制所述网关设备通过所述远端访问地址建立多方语音通话。
  11. 根据权利要求9所述的方法，其特征在于，所述方法还包括：
    所述远端访问地址对应的服务器与主叫侧、指定的终端分别建立数据通道；以及
    控制所述主叫侧和指定的终端之间通过所述数据通道进行的多方语音通话。
  12. 根据权利要求11所述的方法，其特征在于，所述控制所述主叫侧和指定的终端之间通过所述数据通道进行的多方语音通话的步骤包括：
    接收所述多方语音通话中任意网络电话终端发送的数据流或电话终端通过网关设备发送的数据流；以及
    对所述数据流进行转发处理，以使所述数据流通过数据通道传输至相应的终端。
  13. 根据权利要求12所述的方法，其特征在于，接收的所述数据流为多路数据流，所述对所述数据流进行转发处理，以使所述数据流通过数据通道传输至相应的终端的步骤包括：
    确定所述多路数据流的接收端，所述接收端包括网络电话终端和/或电话终端；
    针对每一接收端，所述接收端为网络电话终端且支持混音处理时转发所述多路数据流中所述接收端对应的数据流，所述接收端为电话终端或者所述接收端为网络电话终端且不支持混音处理时将待转发的所述数据流进行混音处理后转发；以及
    所述转发的数据流是任意一路数据流或者任意组合的数据流。
  14. 根据权利要求13所述的方法，其特征在于，对所述电话终端所进行的混音处理包括电话格式的编码，所述转发的数据流是电话终端支持的码流，所述方法还包括：
    所述网关设备接收所述电话终端支持的码流，并将所述码流送达相应的电话终端。
  15. 根据权利要求13所述的方法，其特征在于，所述接收端为电话终端或者所述接收端为网络电话终端且不支持混音处理时将待转发的所述数据流进行混音处理后转发的步骤包括：
    所述接收端为电话终端或者所述接收端为网络电话终端且不支持混音处理时，叠加待转发的所述数据流以输出一路数据流，在输出的所述一路数据流中附带所述接收端的身份标识信息；以及
    向所述接收端转发所述附带接收端的身份标识信息的一路数据流。
  16. 一种即时通话装置，其特征在于，所述装置运行于主叫侧，并且所述主叫侧为网络电话终端，所述装置包括：
    请求发起模块，用于向信令服务器发起多方语音通话请求，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；
    主叫侧通道建立模块，用于根据所述信令服务器响应所述多方语音通话请求返回的远端访问地址建立数据通道；
    通话执行模块，用于待指定的终端根据所述远端访问地址建立数据通道之后进行所述多方语音通话；
    所述终端的数据通道建立中，所述电话终端通过网关设备建立所述远端访问地址对应的数据通道。
  17. 根据权利要求16所述的装置，其特征在于，所述装置还包括请求应答模块和被叫侧通道建立模块，所述请求应答模块和被叫侧通话建立模块运行于指定参与多方语音通话的各网络电话终端中，其中：
    所述请求应答模块用于对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址；
    所述被叫侧通道建立模块用于根据远端访问地址建立所述网络终端的数据通道。
  18. 根据权利要求16或17所述的装置，其特征在于，所述装置还包括运行于所述多方语音通话中网络电话终端的通话接收模块；
    所述通话接收模块用于接收数据通道传输的数据流，并进行自适应混音处理后播放。
  19. 根据权利要求18所述的装置，其特征在于，所述通话接收模块包括：
    接收执行单元，用于通过所述数据通道接收数据流，所述数据流来自其它终端；
    自适应判断单元，用于对所述数据流进行自适应判断，输出所述数据流未进行混音的自适应判断结果；
    本地混音单元，用于根据所述自适应判断结果进行所述数据流的混音处理，并播放混音处理所得到的数据流。
  20. 根据权利要求19所述的装置，其特征在于，所述自适应判断单元进一步用于判断所述数据流是否附带自身的身份标识信息，若为否，则输出所述数据流未进行混音的自适应判断结果，若为是，则输出所述数据流已进行 混音的自适应判断结果，所述自适应判断结果触发进行所述数据流的播放。
  21. 根据权利要求16所述的装置，其特征在于，所述装置还包括：
    数据流输入模块，运行于多方语音通话中的任意网络电话终端，用于编码输入的通话数据得到自身输入的数据流，并通过建立的所述数据通道传输。
  22. 一种即时通话装置，其特征在于，所述装置运行于被叫侧的网络电话终端中，所述装置包括：
    请求应答模块，用于对信令服务器转发的多方语音通话请求进行应答，得到远端访问地址，所述多方语音通话请求中指定参与多方语音通话的终端包括电话终端；
    被叫侧通道建立模块，用于根据所述远端访问地址建立所述网络电话终端的数据通道；
    待所述多方语音通话请求指定的终端根据所述远端访问地址建立数据通道之后进入所述多方语音通话。
  23. 根据权利要求22所述的装置，其特征在于，所述请求应答模块包括：
    转发请求接收单元，用于接收信令服务器为主叫侧转发的多方语音通话请求；
    信令生成单元，用于根据检测的接听操作生成接听应答信令，并将所述接听应答信令传送至所述信令服务器；
    地址接收单元，用于接收所述信令服务器根据所述接听应答信令发送的远端访问地址。
  24. 一种即时通话系统，其特征在于，所述系统包括运行于信令服务器中的请求接收模块、通话请求响应模块和通话建立模块，其中：
    所述请求接收模块用于接收主叫侧发起的多方语音通话请求；
    所述通话请求响应模块用于响应于所述多方语音通话请求，向主叫侧返 回远端访问地址，并向所述多方语音通话请求中指定的终端发起通话建立的信令交互，所述指定的终端包括电话终端；
    所述通话建立模块用于通过所述信令交互向所述指定的终端中包含的网络电话终端和/或所述电话终端对应的网关设备返回远端访问地址，以建立多方语音通话。
  25. 根据权利要求24所述的系统，其特征在于，所述通话建立模块包括：
    请求转换单元，用于为所述电话终端转换所述多方语音通话请求得到电话呼叫请求，并发送到对应的网关设备，所述电话呼叫请求通过所述网关设备送达所述电话终端；
    应答信令接收单元，用于接收网关设备转发的接听应答信令，所述接听应答信令是所述电话终端响应所述电话呼叫请求返回至所述网关设备的；
    信令响应单元，用于响应于所述接听应答信令，向所述网关设备返回远端访问地址，以控制所述网关设备通过所述远端访问地址建立多方语音通话。
  26. 根据权利要求24所述的系统，其特征在于，所述系统还包括通道建立模块和通话控制模块，所述通道建立模块和通话控制模块运行于所述远端访问地址对应的服务器中；
    所述通道建立模块用于与主叫侧、指定的终端分别建立数据通道；
    所述通话控制模块用于控制所述主叫侧和指定的终端之间通过数据通道进行的多方语音通话。
  27. 根据权利要求26所述的系统，其特征在于，所述通话控制模块包括：
    数据流接收单元，用于接收所述多方语音通话中任意网络电话终端发送的数据流或电话终端通过网关设备发送的数据流；
    转发处理单元，用于对所述数据流进行转发处理，以使所述数据流通过数据通道传输至相应的终端。
  28. 根据权利要求27所述的系统，其特征在于，接收的所述数据流为多路数据流，所述转发处理单元包括：
    接收端确定子单元，用于确定所述多路数据流的接收端，所述接收端包括网络电话终端和/或电话终端；
    自适应混音转发子单元，用于当所述接收端为网络电话终端且支持混音处理时，转发所述多路数据流中所述接收端对应的数据流，当所述接收端为电话终端或者所述接收端为网络电话终端且不支持混音处理时，将待转发的所述数据流进行混音处理后转发；
    所述转发的数据流是任意一路数据流或者任意组合的数据流。
  29. 根据权利要求28所述的系统，其特征在于，对所述电话终端所进行的混音处理包括电话格式的编码，所述转发的数据流是电话终端支持的码流，所述系统还包括所述网关设备，所述网关设备用于接收所述电话终端支持的码流，并将所述码流送达相应的电话终端。
  30. 根据权利要求28所述的系统，其特征在于，所述自适应混音转发子单元包括：
    数据流叠加子单元，用当所述接收端为电话终端或者所述接收端为网络电话终端且不支持混音处理时，叠加待转发的所述数据流以输出一路数据流，并在输出的所述一路数据流中附带所述接收端的身份标识信息；
    转发执行子单元，用于向所述接收端转发所述附带接收端的身份标识信息的一路数据流。
  31. 一种即时通话系统，其特征在于，包括参与多方语音通话的终端、信令服务器、数据服务器和网关设备，所述终端包括网络电话终端和电话终端，其中：
    任一网络电话终端向所述信令服务器发起的多方语音通话请求；
    向所述发起多方语音通话请求的网络电话终端返回所述数据服务器的远端访问地址，并向所述多方语音通话的其它终端发起通话建立的信令交互，通过所述信令交互向其它网络电话终端，和/或电话终端对应的网关设备返回远端访问地址；
    所述网络电话终端和/或所述电话终端对应的网关设备通过所述远端访问地址与所述数据服务器建立数据通道，进行多方语音通话。
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