WO2009033401A1 - A communication method, system and service controlling function entity - Google Patents

A communication method, system and service controlling function entity Download PDF

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Publication number
WO2009033401A1
WO2009033401A1 PCT/CN2008/072232 CN2008072232W WO2009033401A1 WO 2009033401 A1 WO2009033401 A1 WO 2009033401A1 CN 2008072232 W CN2008072232 W CN 2008072232W WO 2009033401 A1 WO2009033401 A1 WO 2009033401A1
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WO
WIPO (PCT)
Prior art keywords
terminal
media stream
playing
address
service control
Prior art date
Application number
PCT/CN2008/072232
Other languages
French (fr)
Chinese (zh)
Inventor
Ping Liu
Tianyu Yang
Original Assignee
Huawei Technologies Co., Ltd.
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 Huawei Technologies Co., Ltd. filed Critical Huawei Technologies Co., Ltd.
Publication of WO2009033401A1 publication Critical patent/WO2009033401A1/en

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1069Session establishment or de-establishment
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1101Session protocols
    • H04L65/1104Session initiation protocol [SIP]

Definitions

  • the present invention relates to the field of communications, and in particular, to a communication method, system, and service control function entity. Background technique
  • the Session Initiation Protocol is an important protocol in the Next Generation Net (NGN) and is receiving more and more attention from the communications industry.
  • the SIP protocol is used to solve signaling control in the Internet Protocol (IP) network and communicate with softswitches to provide advanced telephony services across the Internet.
  • IP Internet Protocol
  • PSTN Public Switched Telephone Network
  • the embodiment of the present invention provides a communication method, a system, and a service control function entity, to solve the problem that the screen is in an idle state when the terminal having the capability of playing the media stream does not perform a video call exists in the prior art.
  • a communication method comprising:
  • Receiving a call request from the calling terminal determining, according to the calling terminal identifier and/or the called terminal identifier in the call request, a terminal having a capability of playing the media stream, and acquiring an address of the media stream subscribed by the terminal; The address of the stream is sent to the terminal having the capability to play the media stream.
  • a communication system comprising a calling terminal, a called terminal, and a service control function entity, wherein
  • the calling terminal is configured to initiate a call request to the service control function entity, where the call request includes a calling terminal identifier and/or a called terminal identifier, and if the capability of playing the media stream is performed, according to the calling request
  • the address of the media stream obtained by the service control function entity plays the subscribed media stream;
  • the service control function entity is configured to determine, by the calling terminal identifier and/or the called terminal identifier in the call request, a terminal that has the capability of playing the media stream, and obtain an address of the media stream subscribed by the terminal; Sending an address of the media stream to the terminal having the capability of playing the media stream;
  • the called terminal is configured to: if the device has the capability of playing the media stream, play the subscribed media stream according to the address of the media stream acquired by the service control function entity.
  • a service control function entity includes:
  • a receiving module configured to receive a call request sent by the terminal, where the call request includes a terminal identifier, and a terminal capability determining module, configured to determine whether the terminal corresponding to the terminal identifier has the capability of playing a media stream;
  • An acquiring module configured to acquire an address of a media stream subscribed by the terminal when the terminal corresponding to the terminal identifier has the capability of playing the media stream;
  • a sending module configured to return an address of the media stream to the terminal having the capability of playing a media stream.
  • the terminal when the terminal having the capability of playing the media stream is in the state of not performing the video call, the terminal plays the pre-subscribed media stream data, which improves the application rate of the terminal, and the user experience is better.
  • FIG. 1(a) and 1(b) are schematic diagrams showing the structure of a communication system according to Embodiment 1 of the present invention
  • FIG. 2 is a schematic flowchart showing the steps of a communication process according to Embodiment 2 of the present invention
  • FIG. 3 is a schematic diagram of a signaling process of a communication process according to Embodiment 3 of the present invention.
  • FIG. 4 is a schematic diagram of a signaling process of a communication process according to Embodiment 4 of the present invention.
  • FIG. 5 is a schematic diagram of a signaling process of a communication process according to Embodiment 5 of the present invention.
  • FIG. 6 is a schematic diagram of a signaling process of a communication process according to Embodiment 6 of the present invention.
  • FIG. 7 is a schematic diagram of a signaling flow of implementing a multimedia color vision service after a call is established between a calling terminal and a called terminal according to Embodiment 7 of the present invention
  • Embodiment 8 is a schematic diagram of a signaling process of a communication process according to Embodiment 8 of the present invention.
  • Embodiment 9 is a schematic diagram of a signaling process of a communication process in Embodiment 9 of the present invention.
  • FIG. 10 is a schematic diagram of a signaling process of a communication process in Embodiment 10 of the present invention
  • 11 is a schematic flowchart of signaling process of a communication process according to Embodiment 11 of the present invention
  • 12(a) and 12(b) are schematic diagrams showing a signaling flow of a communication process when both the calling terminal and the called terminal are converged terminals according to Embodiment 12 of the present invention
  • FIG. 13 is a schematic structural diagram of a service control function entity in Embodiment 13 of the present invention.
  • the inventor found that the screen of the SIP videophone terminal is before the call connection between the PSTN terminal and the SIP videophone terminal, or between the SIP videophone terminal and the SIP videophone terminal. Is in an idle state; if a non-video call is made after the call connection is successful, the screen of the SIP videophone terminal is still idle, and the functions of the videophone terminal are not fully applied during these idle periods. The user feels poor.
  • the preset media stream data is played by using the terminal having the media stream playing capability.
  • Terminals having the ability to play media streams include, but are not limited to, SIP videophone terminals, H.323 videophone terminals, and converged terminals.
  • the concept of the converged terminal is: a terminal in which a common PSTN telephone and a videophone (SIP videophone) are combined, the physical entity is one, and the internal logic is divided into a PSTN telephone and a video telephone, and the internal can be set non- Asymmetric Digital Subscriber Line (ADSL) interface and network port.
  • ADSL Asymmetric Digital Subscriber Line
  • the communication system in the first embodiment of the present invention includes a calling terminal 11, a called terminal 12, and a service control function entity 13, wherein the calling terminal 11
  • the call request includes a calling terminal identifier and/or a called terminal identifier, and if the calling terminal 11 has the capability of playing a media stream, playing the service control
  • the media stream corresponding to the address of the media stream subscribed by the calling terminal 11 is obtained by the function entity 13;
  • the service control function entity 13 is configured to determine that the media stream is played according to the calling terminal identifier and/or the called terminal identifier in the call request.
  • the called terminal 12 is configured to play the media stream address of the self-subscription obtained by the service control function entity 13 if it has the capability of playing the media stream.
  • the communication system described in the embodiment of the present invention can implement the multimedia color picture service when the video call is not performed (before the call connection is established or the non-video connection is performed).
  • the system further includes an application server 14 configured to query, by the service control function entity 13, an address of a media stream subscribed by the terminal that plays the media stream capability, and return the address to the service control function entity. 13.
  • the time for playing the media stream by the calling and called terminals having the capability of playing the media stream is also required, including but not limited to the following two situations:
  • the system further includes a timer 15 for starting a timer when the terminal having the capability of playing a media stream starts playing a media stream, and stopping timing when stopping the playing of the media stream. , the final statistics play time.
  • the terminal having the capability of playing a media stream is configured to send a play status to the service control function entity 13 when the media stream is started or stopped; the service control function entity 13 is used.
  • the play status is forwarded to the application server 14; the application server 14 is configured to count the play duration according to the start play event and the stop play event.
  • timer 15 and the application server 14 can also be combined to perform the operation of counting the playing time.
  • FIG. 2 it is a schematic flowchart of the steps of the communication method in the second embodiment of the present invention. As can be seen from the figure, the embodiment mainly includes the following steps:
  • Step 201 The calling terminal initiates a call request by using a service control function entity, where the call request includes a calling terminal identifier and/or a called terminal identifier.
  • the calling terminal At the beginning of the call, the calling terminal first initiates a call request.
  • the call request may also contain other information required for normal calling.
  • the service control function entities include but are not limited to: SIP server (SIP Server), H.323 server (H.323 Server).
  • Step 202 The service control function entity determines whether the master and the called terminal have the capability of playing the media stream according to the identity of the master and the called terminal. If yes, step 203 is performed, and if not, the normal call connection is performed. In this embodiment, the normal PSTN telephone terminal only performs normal calls and does not have the capability of playing media stream data. Therefore, the service control function entity can obtain the home subscriber server (Home Subscriber Server, HSS according to the received terminal identifier. It is determined which one or both of the primary and the called terminal have the ability to play media stream data.
  • Home Subscriber Server Home Subscriber Server
  • Step 203 The service control function entity acquires, from an application server (AS), a media stream address that is subscribed by the calling terminal and/or the called terminal that has the capability of playing the media stream data.
  • AS application server
  • the media stream address includes but is not limited to: a Uniform Resource Locator (URL) of the media stream data.
  • URL Uniform Resource Locator
  • AS includes but is not limited to: Policy Server.
  • Step 204 The service control function entity returns the obtained media stream address to the calling terminal and
  • the media stream that is desired to be played may be pre-ordered, or the telecommunication operator subscribes the media stream data to the terminal, and the corresponding relationship between the calling terminal identifier and the media stream address is established.
  • the service control function entity determines that the calling terminal has the capability of playing the media stream according to the attribute of the calling terminal, the corresponding media stream address is queried from the AS and returned to the calling terminal. The process of determining the capabilities of the called terminal and returning to the media stream address is similar.
  • Step 205 The calling terminal and/or the called terminal having the capability of playing the media stream play the media stream corresponding to the media stream address.
  • the calling terminal and/or the called terminal having the capability of playing media stream are connected to the media stream server according to the address, and play the media stream obtained from the media stream server.
  • the communication method described in the embodiment of the present invention can realize the multimedia color picture service when the video call is not performed (before the call connection is established or the non-video connection is performed).
  • the second embodiment of the present invention further includes the step of the service control function entity connecting the call request to the called terminal.
  • the step may be applied between step 201 and step 202, or may be applied between step 203 and step 205.
  • FIG. 3 it is a schematic diagram of a signaling process for implementing a multimedia color vision service in a communication process according to Embodiment 3 of the present invention.
  • a service control function entity is a SIP Server, and the calling party is configured.
  • the terminal is a SIP videophone terminal, and the media stream address is a URL.
  • the third embodiment mainly includes the following steps:
  • Step 301 The calling terminal picks up the phone and reports the off-hook event to the SIP server.
  • the caller terminal and the SIP server are configured to subscribe to the terminal status report event.
  • the calling terminal is off-hook
  • the status of the calling terminal is reported to the SIP server
  • the off-hook event is reported to the SIP server.
  • the reported end user status can be regarded as a call request
  • the reported information includes the calling terminal identifier and the status information of the calling terminal.
  • Step 302 The SIP Server obtains the URL address of the media stream corresponding to the calling terminal from the policy server.
  • the calling terminal is a SIP videophone terminal, when the calling terminal picks up the phone,
  • the SIP Server queries the calling terminal for the URL address of the media stream that needs to be played, so that the calling terminal plays the media stream when no video call is made.
  • the policy server obtains the URL address of the media stream that the calling terminal needs to play according to a preset manner, including but not limited to:
  • the calling terminal subscribes in advance to the media stream (video clip or video advertisement information, etc.), and establishes a correspondence between the calling terminal identifier and the media stream URL, and the SIP server sends the calling terminal identifier to the policy server, and the policy server is established according to the The corresponding relationship between the calling terminal identifier and the media stream URL finds the URL and returns it to the SIP Server.
  • Step 303 The SIP server sends the received URL to the calling terminal, and the calling terminal plays the media stream corresponding to the URL.
  • the calling terminal acquires the media stream corresponding to the URL from the streaming media server according to the URL, and plays the media stream to the user.
  • the set audio and video data can be played to the user through the terminal as long as the terminal goes off-hook and the video call is not performed.
  • a calling terminal is a PSTN terminal
  • a called terminal is a SIP videophone terminal.
  • the calling terminal listens to the ring back tone and the called terminal is ringing, the called terminal plays the media stream data.
  • This process includes: Step 401: The calling terminal initiates a call request through the SIP server.
  • the calling terminal sends a call request to the SIP server, where the call request includes the calling terminal identifier and the called terminal identifier.
  • the SIP message is taken as an example to describe the implementation of the color video service. Therefore, the call request may be an invite message (SIP INVITE).
  • Step 402 The SIP Server determines that the called terminal has the capability of playing the media stream.
  • the SIP Server determines that the calling terminal is a PSTN terminal and the called terminal is a SIP videophone terminal according to the calling terminal identifier and the called terminal identifier in the received call request.
  • Step 403 The SIP server acquires a media stream address corresponding to the called terminal from the policy server.
  • the policy server finds the address of the subscribed media stream according to the called terminal identifier, and returns it to the SIP.
  • Step 404 The SIP Server sends the call request to the called terminal to connect the call.
  • the SIP Server After receiving the returned media stream address, the SIP Server connects the call between the calling terminal and the called terminal: sends a call request to the called terminal, and receives the SIP 180 response returned by the called terminal.
  • the calling terminal hears the ring back tone, and the called terminal starts ringing.
  • Step 405 The SIP server sends the media stream address to the called terminal, and the called terminal plays the corresponding media stream according to the media stream address.
  • the calling terminal when the calling terminal is a SIP videophone terminal and the called terminal is a PSTN terminal, the calling terminal can also play the preset media stream, as shown in FIG. 5, which is implemented by the present invention.
  • the signaling flow diagram of the communication process in the fifth embodiment is different from the fourth embodiment in that: the SIP server determines that the calling terminal has the capability of playing the media stream, and sends the media stream address obtained from the policy server to the calling terminal. The media stream is obtained and played by the calling terminal from the streaming server according to the media stream address.
  • the communication process can be implemented according to the signaling flow diagram shown in FIG. 6 corresponding to the sixth embodiment of the present invention.
  • the SIP server determines that both the calling terminal and the called terminal are SIP videophone terminals, and then query the address of the media stream corresponding to the two terminals from the policy server, and the calling terminal hears the ring back tone, and the called party After the terminal rings, the media stream is obtained and played from the streaming server according to the address of the media stream.
  • the process of implementing the color vision service when the call between the calling terminal and the called terminal is not in the call, and the subsequent embodiments in the seventh, eighth, and ninth are at the calling terminal and the called party.
  • the process of implementing the color vision service after the call between the terminals is established.
  • the terminal is a SIP videophone terminal, including:
  • Step 701 The calling terminal sends a call request to the SIP server, and the SIP server connects the call request to the called terminal to establish a call connection between the calling terminal and the called terminal.
  • the call request includes a calling terminal identifier and a called terminal identifier.
  • the SIP Server After receiving the call request, the SIP Server connects the call to the called terminal according to the called terminal identifier: the SIP Server sends the SIP INVITE to the called terminal, and The SIP 180 response message returned by the called terminal is received. At this time, the calling terminal hears the ring back tone, and the called terminal rings.
  • the called terminal picks up the phone, it sends a SIP 200 response message to the SIP Server. After the SIP Server returns the response message (SIP ACK) to the called terminal, the call between the primary and the called terminal starts.
  • SIP ACK response message
  • Step 702 The SIP server acquires an address of the media stream corresponding to the called terminal from the policy server.
  • the SIP server determines that the called terminal is a SIP videophone terminal according to the called terminal identifier, and the policy server queries the address of the media stream customized for the called terminal.
  • Step 703 After receiving the media stream address, the called terminal plays the media stream corresponding to the media stream address.
  • the calling terminal is a SIP videophone and the called terminal is a PSTN terminal, which is similar to the basic process of the seventh embodiment. The difference is that after the calling terminal and the called terminal establish a connection, the SIP server The address of the media stream subscribed to the calling terminal is queried from the policy server, and the calling terminal obtains and plays the corresponding media stream from the streaming media server according to the address of the media stream, as shown in FIG. 8.
  • the calling terminal and the called terminal are both SIP videophone terminals.
  • This embodiment can be regarded as a combination of Embodiment 7 and Embodiment 8, as shown in FIG.
  • the SIP server obtains the addresses of the media streams customized by the calling terminal and the called terminal respectively from the policy server, and the calling terminal and the called terminal respectively obtain and play from the streaming media server according to the address of the media stream. Media stream.
  • the duration of playing the media stream by the calling and called terminals may be counted, and the duration information is returned to the user.
  • the method of counting the duration includes, but is not limited to, starting a timer when the terminal starts playing the media stream, stopping the timer when the media stream is stopped, and obtaining the playing duration according to the timer; and subscribing between the terminal having the capability of playing the media stream and the SIP Server
  • the terminal status reports the event.
  • the broadcast status can be sent to the SIP server through a message such as SIP.
  • the SIP server transmits the play status to the AS.
  • the AS counts the play duration according to the start play event and the stop play event. .
  • the master and the called terminal belong to the same SIP server.
  • the master and the called terminal may belong to different SIP servers.
  • FIG. 10 it is a schematic diagram of a signaling process in the tenth embodiment of the present invention, where the calling terminal and the called terminal belong to different SIP servers.
  • the dotted line box identified by 1001 in the figure indicates the process in which the called terminal plays the media stream when ringing; the dotted line box identified by 1002 in the figure indicates the process in which the calling terminal plays the media stream while listening to the ring back tone;
  • the dashed box of the logo indicates the process of playing the media stream when the calling terminal and the called terminal perform a non-video call.
  • the application server returns the address of the media stream subscribed by the called terminal to the SIP server serving the called terminal, and may also forward the address by the SIP server serving the calling terminal.
  • the implementation of the color-vision service of the third embodiment to the sixth embodiment is implemented when the call connection is not established between the calling and the called terminals.
  • the implementation of the color-vision service of the seventh embodiment to the tenth embodiment is to establish a non-connection between the calling and called terminals.
  • After the video call is connected in the embodiment of the present invention, including but not limited to: the process before the call connection is established and the process after the call connection is established to implement the color vision service.
  • FIG. 11 is a schematic diagram of a signaling flow according to Embodiment 11 of the present invention.
  • both the primary and the called terminals are SIP videophone terminals
  • the service control function entity is a home domain server node (S- CSCF), before the call connection, respectively play the subscribed media stream, after the non-video call, the main and called terminals still play the subscribed media stream.
  • S- CSCF home domain server node
  • Step 1101 The calling terminal initiates a call request through the S-CSCF, where the call request may include identifiers of the calling and called terminals.
  • the call request may be initiated after the calling terminal picks up the phone, and the call request may be an INVITE message, which is sent to the S-CSCF through the proxy node (P-CSCF), and the S-CSCF determines that the calling and called terminals have the playing media stream. Capabilities, then forward the call request to the application server.
  • Step 1102 The S-CSCF obtains, from the application server, a media stream address corresponding to the primary and the called terminal respectively.
  • the application server queries the address of the subscribed media stream according to the primary and called terminal identifiers, and returns the address to the S-CSCF through the INVITE.
  • Step 1103 The S-CSCF sends the call request to the called terminal to connect the call.
  • the S-CSCF that provides the service to the calling terminal forwards the call request to the called terminal through the S-CSCF that provides the service to the called terminal. After the called side returns the 180 response message, the calling terminal is in the state of listening to the ring back tone. The user is called to be ringing.
  • Step 1104 The S-CSCF sends the media stream address to the calling terminal and the called terminal side, and the primary and the called terminal acquires the media stream corresponding to the address, and plays the media stream.
  • the S-CSCF serving the calling terminal sends the media stream address obtained from the application server to the calling terminal and the called terminal through the P-CSCF and the S-CSCF serving the called terminal respectively, the master and the called party.
  • the terminal acquires the corresponding media stream from the streaming media server according to the address, and plays the content to the user.
  • the address may also be sent by the application server to the calling and called terminals.
  • Step 1105 The called terminal picks up the phone, and the call connection is established.
  • the application server can query the media stream addresses subscribed by the calling and called terminals again according to the identifiers of the calling and called terminals (the primary and the called terminals can subscribe to different media streams at different times).
  • Step 1106 The application server sends the address to the calling terminal and the called terminal, and the primary and the called terminal obtain the corresponding media stream according to the address, and play the same.
  • step 1105 If the application server does not query the media stream address again in step 1105, then the media stream is played in accordance with the address obtained in step 1104 in this step.
  • the SIP protocol is used as the data transmission protocol
  • the terminal having the capability of playing the media stream is a SIP videophone terminal.
  • the setting has a play.
  • the terminal of the media stream capability is a converged terminal, and both the calling terminal and the called terminal are converged terminals, and the communication process is shown in FIG. 12(a) and FIG. 12(b), and FIG. 12(a) shows the calling terminal and the called party.
  • the color vision service process is implemented when the call connection is not established between the terminals, including:
  • Step 1201 The PSTN terminal in the calling terminal initiates a call request to the PSTN terminal in the called terminal by using the SIP server, where the call request includes the calling terminal identifier and the called terminal identifier.
  • the PSTN in the calling terminal initiates a request to the PSTN in the called terminal in the normal manner.
  • the PSTN terminal in the calling terminal sends the call request to the SIP Server A that provides the service to the calling terminal through the initial address message (IMM), and the SIP Server A forwards the call request to the called terminal through the INVITE message.
  • SIP Server B SIP Server B sends the call request again through the IAM to the PSTN terminal of the called terminal.
  • the PSTN terminal in the called terminal returns the response response message to the PSTN terminal in the calling terminal.
  • Step 1202 The SIP Server determines that the calling and called terminals have the ability to play the media stream, and query the address of the media stream customized by the calling and called terminals from the policy server.
  • the SIP Server obtains the address of the media stream from the policy server and can carry the required information (such as the called terminal identifier, the calling terminal identifier, and the call information, etc.) through the information (INFO) message.
  • the required information such as the called terminal identifier, the calling terminal identifier, and the call information, etc.
  • Step 1203 The policy server returns the queried media stream address to the primary SIP server that is served by the terminal.
  • the returned media stream address may be carried by the 200 response message.
  • Step 1204 The SIP server returns the media stream address to the SIP terminal in the calling and called terminals respectively.
  • the media stream address corresponding to the calling terminal is set to URL A1
  • the media stream address corresponding to the called terminal is URL B1
  • the SIP server returns the URL A1 to the SIP terminal in the calling terminal by using the INFO message, and returns the URL B1.
  • Step 1205 The primary and the called terminal respectively receive the media stream address to acquire and play the media stream.
  • the primary and secondary terminals return a 200 response message to the SIP Server that provides services for itself.
  • the color picture service before the call is established between the primary and the called terminal is implemented.
  • the color vision service after the call is established between the primary and the called terminal including:
  • Step 1206 Establish a normal call connection between the calling terminal and the called terminal.
  • the called terminal picks up the phone, and sends the information to the SIP server that provides the server for the called terminal through the response message (ANM), and the SIP server forwards the information to the SIP server of the server provided by the calling terminal through the 200 message, the SIP Server.
  • the information is sent to the PSTN terminal of the calling terminal through the ANM message, and the ACK message is returned to the SIP Server serving the called terminal.
  • Steps 1207 through 1210 are similar to steps 1202 through 1205 described above.
  • Step 1211 The SIP terminal of the calling terminal forwards the INFO message carrying the playing duration to the policy server through the SIP server serving the self.
  • the process in which the called terminal returns the playing duration is similar.
  • the SIP terminal of the called terminal also forwards the INFO message carrying the playing duration to the policy server through the SIP server that provides the service itself.
  • the Policy Server returns a 200 response message over the same path.
  • one or two of the calling terminal and the called terminal are required to have the capability of playing the media stream, and the type of the terminal itself is not limited, and may be a SIP terminal, H. 323 terminal or converged terminal or separate terminal, therefore, the communication process may be communication between different types of terminals, and is not limited to the case described in the foregoing embodiment.
  • the SIP server When the SIP server requests the policy server to query the media stream address, the information can be sent through INFO, MESSAGE, etc.; the policy server returns the obtained media stream address to the SIP server through the 200, INFO, MESSAGE and other messages.
  • the thirteenth embodiment of the present invention further provides a service control function entity.
  • the service control function entity includes a receiving module 21, a terminal capability determining module 22, and an obtaining module 23.
  • the receiving module 21 is configured to receive a call request sent by the terminal, where the call request includes a terminal identifier
  • the terminal capability determining module 22 is configured to determine whether the terminal corresponding to the terminal identifier has the capability of playing a media stream.
  • the address of the media stream subscribed by the terminal is obtained.
  • the service control function entity further includes a sending module 24, configured to return an address of the media stream to the terminal having the capability of playing a media stream.
  • the sending module 24 may carry the media stream address in an INFO message and send it to the terminal.
  • the SIP message is used as a carrier for transmission.
  • Information if in the H.323 protocol, transmits the required information using the H.323 message as the carrier.
  • the SIP message generally supports the TEL URI/SIP URI.
  • the SIP message needs to carry the RTSP (Real-Time Streaming Protocol) URL or HTTP (for the media stream in the media stream server).
  • the INFO message is the format of the transmitted message, and the cmd command word indicates the action to be performed.
  • the cmd is 4001, indicating that the terminal having the media stream capability plays the subscribed media stream when the calling terminal and the called terminal are in the process of making a video call. .
  • ⁇ linkuri> indicates the address of the media stream.
  • the address type can be an HTTP URL or an RTSP URL.
  • the address type is an HTTP URL
  • the terminal is linked to the corresponding website.
  • the address type is RTSP URL
  • the terminal is linked to the corresponding streaming server. .
  • ⁇ cont> indicates the attached description of the linkuri link.
  • ⁇ event> indicates a status event in which the terminal starts, stops, and pauses the playback of the media stream.
  • ⁇ autolink> indicates the terminal connection method, either manually or automatically. For example, if autolink is 2, the terminal automatically connects to the streaming server after receiving the message, and the terminal does not need any operation.
  • the SIP server and the application server also need to implement signaling transmission.
  • the SIP protocol is used, the SIP message is used as the carrier to transmit the required information.
  • the H.323 protocol is used, the H.323 message is used as the carrier transmission.
  • Information for example, the SIP message generally supports the TEL URI/SIP URI.
  • the SIP message needs to carry the RTSP URL or the HTTP URL of the media stream in the media stream server, so the URI address in the SIP message needs to be extended. Make SIP messages also support RTSP
  • Table 1 and Table 2 are the interface information of the INFO message, as shown in Table 3, the interface information of the 200 message ⁇
  • cmd, linkuri, cont, event, autolink Tag parameters are the same as softswitch and terminal interface. Meaning.
  • ⁇ callee> indicates the called terminal identity
  • ⁇ caller> indicates the calling terminal identity
  • ⁇ callinfo> indicates normal call time information
  • the method, system and service control function entity of the present invention play pre-subscribed media stream data when the main and called terminals do not make a video call, so that the user can obtain more information at the same time, and the experience is better.
  • the spirit and scope of the Ming Thus, it is intended that the present invention cover the modifications and variations of the inventions

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Abstract

A communication method includes: calling terminal originates a call request to service controlling function entity, and the call request includes the calling terminal's identifier and/or the called terminal's identifier; the service controlling function entity determines the terminal capable of playing media stream according to the calling terminal's identifier and/or the called terminal's identifier, and obtains the address of the media stream which the terminal has subscribed; the terminal capable of playing media stream obtains and plays the subscribed media stream according to the media stream address. A communication system and service controlling function entity are also provided.

Description

一种通信方法、 系统和业务控制功能实体  Communication method, system and service control function entity
本申请要求于 2007 年 9 月 13 日提交中国专利局、 申请号为 200710153905.2、 发明名称为"一种通信方法、 系统和业务控制功能实体"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。  This application claims priority to Chinese Patent Application No. 200710153905.2, entitled "Communication Method, System and Service Control Function Entity", filed on September 13, 2007, the entire contents of which are incorporated by reference. In this application.
技术领域 Technical field
本发明涉及通信领域,尤其涉及一种通信方法、系统和业务控制功能实体。 背景技术  The present invention relates to the field of communications, and in particular, to a communication method, system, and service control function entity. Background technique
会话发起协议 ( Session Initiation Protocol , SIP )是下一代网络 ( Next Generation Net, NGN )中的重要协议, 越来越得到通信业界的重视。 SIP协议 用来解决因特网协议(Internet Protocol, IP ) 网中的信令控制, 并同软交换进 行通信, 提供跨越因特网 (Internet ) 的高级电话业务。  The Session Initiation Protocol (SIP) is an important protocol in the Next Generation Net (NGN) and is receiving more and more attention from the communications industry. The SIP protocol is used to solve signaling control in the Internet Protocol (IP) network and communicate with softswitches to provide advanced telephony services across the Internet.
电信网络中存在大量的公共电话交换网 ( Public Switched Telephone Network, PSTN )用户, 随着电信的发展, 使用 SIP视频电话的用户也越来越 多, PSTN终端与 SIP可视电话终端之间, 或 SIP可视电话终端与 SIP可视电 话终端之间的通话连接之前, SIP可视电话终端的屏幕都是处于闲置状态; 在 通话连接成功后若进行的是非视频通话,则 SIP可视电话终端的屏幕仍然是处 于闲置状态。  There are a large number of Public Switched Telephone Network (PSTN) users in the telecommunication network. With the development of telecommunication, more and more users use SIP video phones, between PSTN terminals and SIP video phone terminals, or Before the call connection between the SIP videophone terminal and the SIP videophone terminal, the screen of the SIP videophone terminal is in an idle state; if the non-video call is performed after the call connection is successful, the SIP videophone terminal The screen is still idle.
发明内容 Summary of the invention
本发明实施例提供一种通信方法、 系统和业务控制功能实体, 以解决现有 技术中存在的在利用具有播放媒体流能力的终端未进行视频通话时,屏幕处于 闲置状态的问题。  The embodiment of the present invention provides a communication method, a system, and a service control function entity, to solve the problem that the screen is in an idle state when the terminal having the capability of playing the media stream does not perform a video call exists in the prior art.
一种通信方法, 该方法包括:  A communication method, the method comprising:
接收来自主叫终端的呼叫请求, 根据所述呼叫请求中的主叫终端标识和 / 或被叫终端标识确定具有播放媒体流能力的终端,获取该终端订阅的媒体流的 地址; 将所述媒体流的地址发送至所述具有播放媒体流能力的终端。  Receiving a call request from the calling terminal, determining, according to the calling terminal identifier and/or the called terminal identifier in the call request, a terminal having a capability of playing the media stream, and acquiring an address of the media stream subscribed by the terminal; The address of the stream is sent to the terminal having the capability to play the media stream.
一种通信系统, 该系统包括主叫终端、被叫终端和业务控制功能实体, 其 中,  A communication system, the system comprising a calling terminal, a called terminal, and a service control function entity, wherein
所述主叫终端, 用于向所述业务控制功能实体发起呼叫请求,该呼叫请求 中包含主叫终端标识和 /或被叫终端标识, 若具有播放媒体流的能力, 则根据 所述业务控制功能实体获取的媒体流的地址播放订阅的媒体流; The calling terminal is configured to initiate a call request to the service control function entity, where the call request includes a calling terminal identifier and/or a called terminal identifier, and if the capability of playing the media stream is performed, according to the calling request The address of the media stream obtained by the service control function entity plays the subscribed media stream;
所述业务控制功能实体, 用于 ^居所述呼叫请求中的主叫终端标识和 /或 被叫终端标识确定具有播放媒体流能力的终端,并获取该终端订阅的媒体流的 地址; 将所述媒体流的地址发送至所述具有播放媒体流能力的终端;  The service control function entity is configured to determine, by the calling terminal identifier and/or the called terminal identifier in the call request, a terminal that has the capability of playing the media stream, and obtain an address of the media stream subscribed by the terminal; Sending an address of the media stream to the terminal having the capability of playing the media stream;
所述被叫终端, 用于若具有播放媒体流的能力, 则根据所述业务控制功能 实体获取的媒体流的地址播放订阅的媒体流。  The called terminal is configured to: if the device has the capability of playing the media stream, play the subscribed media stream according to the address of the media stream acquired by the service control function entity.
一种业务控制功能实体, 该业务控制功能实体包括:  A service control function entity, the service control function entity includes:
接收模块, 用于接收终端发送的呼叫请求, 该呼叫请求中包含终端标识; 终端能力确定模块,用于确定所述终端标识对应的终端是否具有播放媒体 流的能力;  a receiving module, configured to receive a call request sent by the terminal, where the call request includes a terminal identifier, and a terminal capability determining module, configured to determine whether the terminal corresponding to the terminal identifier has the capability of playing a media stream;
获取模块, 用于当所述终端标识对应的终端具有播放媒体流能力时,获取 所述终端订阅的媒体流的地址;  An acquiring module, configured to acquire an address of a media stream subscribed by the terminal when the terminal corresponding to the terminal identifier has the capability of playing the media stream;
发送模块,用于将所述媒体流的地址返回给具有播放媒体流能力的所述终 端。  And a sending module, configured to return an address of the media stream to the terminal having the capability of playing a media stream.
本发明实施例中,当具有播放媒体流能力的终端摘机后处于未进行视频通 话的状态时, 该终端播放预先订阅的媒体流数据, 提高了终端的应用率, 使用 户体验较好。  In the embodiment of the present invention, when the terminal having the capability of playing the media stream is in the state of not performing the video call, the terminal plays the pre-subscribed media stream data, which improves the application rate of the terminal, and the user experience is better.
附图说明 DRAWINGS
图 1 ( a )和图 1 ( b )为本发明实施例一中通信系统的结构示意图; 图 2为本发明实施例二中通信过程的步骤流程示意图;  1(a) and 1(b) are schematic diagrams showing the structure of a communication system according to Embodiment 1 of the present invention; FIG. 2 is a schematic flowchart showing the steps of a communication process according to Embodiment 2 of the present invention;
图 3为本发明实施例三中通信过程的信令流程示意图;  3 is a schematic diagram of a signaling process of a communication process according to Embodiment 3 of the present invention;
图 4为本发明实施例四中通信过程的信令流程示意图;  4 is a schematic diagram of a signaling process of a communication process according to Embodiment 4 of the present invention;
图 5为本发明实施例五中通信过程的信令流程示意图;  5 is a schematic diagram of a signaling process of a communication process according to Embodiment 5 of the present invention;
图 6为本发明实施例六中通信过程的信令流程示意图;  6 is a schematic diagram of a signaling process of a communication process according to Embodiment 6 of the present invention;
图 7为本发明实施例七中,在主叫终端和被叫终端之间建立通话后实现多 媒体彩视业务的信令流程示意图;  7 is a schematic diagram of a signaling flow of implementing a multimedia color vision service after a call is established between a calling terminal and a called terminal according to Embodiment 7 of the present invention;
图 8为本发明实施例八中通信过程的信令流程示意图;  8 is a schematic diagram of a signaling process of a communication process according to Embodiment 8 of the present invention;
图 9为本发明实施例九中通信过程的信令流程示意图;  9 is a schematic diagram of a signaling process of a communication process in Embodiment 9 of the present invention;
图 10为本发明实施例十中通信过程的信令流程示意图; 图 11为本发明实施例十一中通信过程的信令流程示意图; 10 is a schematic diagram of a signaling process of a communication process in Embodiment 10 of the present invention; 11 is a schematic flowchart of signaling process of a communication process according to Embodiment 11 of the present invention;
图 12 ( a )和图 12 ( b )为本发明实施例十二中, 主叫终端和被叫终端都 为融合终端时通信过程的信令流程示意图;  12(a) and 12(b) are schematic diagrams showing a signaling flow of a communication process when both the calling terminal and the called terminal are converged terminals according to Embodiment 12 of the present invention;
图 13为本发明实施例十三中业务控制功能实体结构示意图。  FIG. 13 is a schematic structural diagram of a service control function entity in Embodiment 13 of the present invention.
具体实施方式 detailed description
在进行本发明创造过程中,发明人发现由于 PSTN终端与 SIP可视电话终 端之间, 或 SIP可视电话终端与 SIP可视电话终端之间的通话连接之前, SIP 可视电话终端的屏幕都是处于闲置状态;在通话连接成功后若进行的是非视频 通话, 则 SIP可视电话终端的屏幕仍然是处于闲置状态, 在这些处于闲置的时 间可视电话终端的功能没有得到充分的应用, 使用户感受较差。  In carrying out the creation process of the present invention, the inventor found that the screen of the SIP videophone terminal is before the call connection between the PSTN terminal and the SIP videophone terminal, or between the SIP videophone terminal and the SIP videophone terminal. Is in an idle state; if a non-video call is made after the call connection is successful, the screen of the SIP videophone terminal is still idle, and the functions of the videophone terminal are not fully applied during these idle periods. The user feels poor.
本发明实施例中,在未进行视频通话时, 利用具有媒体流播放能力的终端 来播放预设的媒体流数据。  In the embodiment of the present invention, when the video call is not performed, the preset media stream data is played by using the terminal having the media stream playing capability.
所谓具有播放媒体流能力的终端包括但不限于: SIP可视电话终端、 H.323 可视电话终端和融合终端。 融合终端的概念为: 普通 PSTN话机和可视电话的 话机(SIP可视电话话机)融合在一起的一种终端, 其物理实体是一个, 而内 部逻辑分成 PSTN 话机和视频话机, 内部可以设置非对称数字用户线 ( Asymmetric Digital Subscriber Line , ADSL )接口和网口。  Terminals having the ability to play media streams include, but are not limited to, SIP videophone terminals, H.323 videophone terminals, and converged terminals. The concept of the converged terminal is: a terminal in which a common PSTN telephone and a videophone (SIP videophone) are combined, the physical entity is one, and the internal logic is divided into a PSTN telephone and a video telephone, and the internal can be set non- Asymmetric Digital Subscriber Line (ADSL) interface and network port.
下面结合说明书附图详细描述本发明。  The invention will be described in detail below with reference to the accompanying drawings.
如图 1 ( a )和图 1 ( b )所示, 为本发明实施例一中通信系统, 该系统包 括主叫终端 11、 被叫终端 12和业务控制功能实体 13 , 其中, 主叫终端 11用 于向所述业务控制功能实体 13发起呼叫请求, 该呼叫请求中包含主叫终端标 识和 /或被叫终端标识, 若主叫终端 11 自身具有播放媒体流的能力, 则播放所 述业务控制功能实体 13获取的主叫终端 11订阅的媒体流的地址对应的媒体 流; 业务控制功能实体 13用于根据所述呼叫请求中的主叫终端标识和 /或被叫 终端标识确定具有播放媒体流能力的终端, 并获取该终端订阅的媒体流的地 址; 被叫终端 12用于若自身具有播放媒体流的能力, 则播放所述业务控制功 能实体 13获取的自身订阅的媒体流的地址对应的媒体流。  As shown in FIG. 1(a) and FIG. 1(b), the communication system in the first embodiment of the present invention includes a calling terminal 11, a called terminal 12, and a service control function entity 13, wherein the calling terminal 11 For initiating a call request to the service control function entity 13, the call request includes a calling terminal identifier and/or a called terminal identifier, and if the calling terminal 11 has the capability of playing a media stream, playing the service control The media stream corresponding to the address of the media stream subscribed by the calling terminal 11 is obtained by the function entity 13; the service control function entity 13 is configured to determine that the media stream is played according to the calling terminal identifier and/or the called terminal identifier in the call request. The terminal of the capability, and obtains the address of the media stream subscribed by the terminal; the called terminal 12 is configured to play the media stream address of the self-subscription obtained by the service control function entity 13 if it has the capability of playing the media stream. Media stream.
在本发明实施例中所描述的通信系统除了进行通话连接之外 ,还可以在未 进行视频通话时(通话连接建立之前或进行非视频连接)实现多媒体彩视业务。 另外, 所述系统还包括应用服务器 14, 用于查询出所述业务控制功能实 体 13确定的具有播放媒体流能力的终端订阅的媒体流的地址, 并将该地址返 回给所述业务控制功能实体 13。 In addition to the call connection, the communication system described in the embodiment of the present invention can implement the multimedia color picture service when the video call is not performed (before the call connection is established or the non-video connection is performed). In addition, the system further includes an application server 14 configured to query, by the service control function entity 13, an address of a media stream subscribed by the terminal that plays the media stream capability, and return the address to the service control function entity. 13.
为了方便用户使用本系统 ,在本实施例中还需要统计具有播放媒体流的能 力的主、 被叫终端播放媒体流的时间, 包括但不限以下两种情况:  In order to facilitate the user to use the system, in this embodiment, the time for playing the media stream by the calling and called terminals having the capability of playing the media stream is also required, including but not limited to the following two situations:
1、 如图 1 ( a )所示, 所述系统还包括计时器 15, 用于在具有播放媒体流 的能力的所述终端开始播放媒体流时启动计时器,在停止播放媒体流时停止计 时器, 最终统计播放时长。  1. As shown in FIG. 1(a), the system further includes a timer 15 for starting a timer when the terminal having the capability of playing a media stream starts playing a media stream, and stopping timing when stopping the playing of the media stream. , the final statistics play time.
2、 如图 1 ( b )所示, 具有播放媒体流的能力的所述终端, 用于在启动或 停止播放媒体流时将播放状态发送给业务控制功能实体 13; 业务控制功能实 体 13用于将所述播放状态转发给应用服务器 14; 应用服务器 14用于根据启 动播放事件和停止播放事件统计播放时长。  2. As shown in FIG. 1(b), the terminal having the capability of playing a media stream is configured to send a play status to the service control function entity 13 when the media stream is started or stopped; the service control function entity 13 is used. The play status is forwarded to the application server 14; the application server 14 is configured to count the play duration according to the start play event and the stop play event.
另夕卜,还可以将计时器 15和应用服务器 14结合在一起,执行统计播放时 长的操作。  In addition, the timer 15 and the application server 14 can also be combined to perform the operation of counting the playing time.
在本发明实施例的系统和后续所描述的方法中,都可以应用于 SIP协议或 In the system of the embodiment of the present invention and the method described in the following, both can be applied to the SIP protocol or
H.323协议等。 H.323 protocol, etc.
下面结合具体实施例详细描述本发明方法。  The method of the present invention is described in detail below in conjunction with specific embodiments.
主叫终端或被叫终端中的一个或两个具有播放媒体流数据的能力。 如图 2 所示, 为本发明实施例二中通信方法步骤流程示意图, 从图中可以看出, 本实 施例主要包括以下步骤:  One or both of the calling or called terminals have the ability to play media stream data. As shown in FIG. 2, it is a schematic flowchart of the steps of the communication method in the second embodiment of the present invention. As can be seen from the figure, the embodiment mainly includes the following steps:
步骤 201 : 主叫终端通过业务控制功能实体发起呼叫请求, 该呼叫请求中 包含了主叫终端标识和 /或被叫终端标识。  Step 201: The calling terminal initiates a call request by using a service control function entity, where the call request includes a calling terminal identifier and/or a called terminal identifier.
在呼叫开始时主叫终端首先发起呼叫请求,该呼叫请求中除了包含主叫终 端标识和 /或被叫终端标识之外, 还可以包含了其他正常呼叫时需要的信息。  At the beginning of the call, the calling terminal first initiates a call request. In addition to the calling terminal identifier and/or the called terminal identifier, the call request may also contain other information required for normal calling.
业务控制功能实体包括但不限于: SIP服务器(SIP Server ), H.323服务 器( H.323 Server )。  The service control function entities include but are not limited to: SIP server (SIP Server), H.323 server (H.323 Server).
步骤 202: 业务控制功能实体根据主、 被叫终端标识分别判断主、 被叫终 端是否具有播放媒体流的能力, 若具有, 则执行步骤 203 , 若都不具有, 则进 行正常的通话连接。 在本实施例中, 普通的 PSTN电话终端只进行正常的通话, 并不具有播放 媒体流数据的能力,因此业务控制功能实体可以根据接收到的终端标识从归属 签约用户服务器( Home Subscriber Server, HSS )确定主、 被叫终端中哪一个 或两个具有播放媒体流数据的能力。 Step 202: The service control function entity determines whether the master and the called terminal have the capability of playing the media stream according to the identity of the master and the called terminal. If yes, step 203 is performed, and if not, the normal call connection is performed. In this embodiment, the normal PSTN telephone terminal only performs normal calls and does not have the capability of playing media stream data. Therefore, the service control function entity can obtain the home subscriber server (Home Subscriber Server, HSS according to the received terminal identifier. It is determined which one or both of the primary and the called terminal have the ability to play media stream data.
步骤 203: 业务控制功能实体从应用服务器(Application Server, AS )处 获取具有播放媒体流数据能力的主叫终端和 /或被叫终端订阅的媒体流地址。  Step 203: The service control function entity acquires, from an application server (AS), a media stream address that is subscribed by the calling terminal and/or the called terminal that has the capability of playing the media stream data.
所述媒体流地址包括但不限于: 媒体流数据的统一资源定位符(Universal Resource Locator, URL )。  The media stream address includes but is not limited to: a Uniform Resource Locator (URL) of the media stream data.
AS包括但不限于: 策略服务器。  AS includes but is not limited to: Policy Server.
步骤 204: 业务控制功能实体将获取的媒体流地址返回给所述主叫终端和 Step 204: The service control function entity returns the obtained media stream address to the calling terminal and
/或被叫终端。 / or called terminal.
若主叫终端具有播放媒体流的能力,可以预先订制希望播放的媒体流,或 由电信运行商为终端订制媒体流数据,则建立主叫终端标识与媒体流地址的对 应关系。当业务控制功能实体根据主叫终端的属性确定主叫终端具有播放媒体 流的能力时, 从 AS中查询出对应的媒体流地址, 并返回给主叫终端。 判断被 叫终端的能力并返回媒体流地址的过程类似。  If the calling terminal has the capability of playing the media stream, the media stream that is desired to be played may be pre-ordered, or the telecommunication operator subscribes the media stream data to the terminal, and the corresponding relationship between the calling terminal identifier and the media stream address is established. When the service control function entity determines that the calling terminal has the capability of playing the media stream according to the attribute of the calling terminal, the corresponding media stream address is queried from the AS and returned to the calling terminal. The process of determining the capabilities of the called terminal and returning to the media stream address is similar.
步骤 205:具有播放媒体流能力的主叫终端和 /或被叫终端播放所述媒体流 地址对应的媒体流。  Step 205: The calling terminal and/or the called terminal having the capability of playing the media stream play the media stream corresponding to the media stream address.
具有播放媒体流能力的主叫终端和 /或被叫终端接收到所述媒体流地址之 后 ,根据该地址连接到媒体流服务器 ,并播放从媒体流服务器处获得的媒体流。  After receiving the media stream address, the calling terminal and/or the called terminal having the capability of playing media stream are connected to the media stream server according to the address, and play the media stream obtained from the media stream server.
在本发明实施例中所描述的通信方法除了进行通话连接之外,还可以在未 进行视频通话时(通话连接建立之前或进行非视频连接)实现多媒体彩视业务。  In addition to the call connection, the communication method described in the embodiment of the present invention can realize the multimedia color picture service when the video call is not performed (before the call connection is established or the non-video connection is performed).
本发明实施例二中还包括业务控制功能实体将呼叫请求接续到被叫终端 的步骤, 该步骤可以应用在步骤 201与步骤 202之间, 也可以应用在步骤 203 和步骤 205之间。  The second embodiment of the present invention further includes the step of the service control function entity connecting the call request to the called terminal. The step may be applied between step 201 and step 202, or may be applied between step 203 and step 205.
在呼叫接续之前, 利用 SIP可视电话终端播放音、视频数据时, 根据主被 叫终端的种类存在多种情况, 下面针对不同的情况分别进行描述。  Before the call is connected, when the audio and video data are played by the SIP videophone terminal, there are various cases depending on the type of the called terminal. The following describes each case separately.
如图 3所示,为本发明实施例三中在通信过程中实现多媒体彩视业务的信 令流程示意图, 在本实施例三中, 设定业务控制功能实体为 SIP Server, 主叫 终端为 SIP可视电话终端, 媒体流地址为 URL, 本实施例三主要包括以下步 骤: As shown in FIG. 3, it is a schematic diagram of a signaling process for implementing a multimedia color vision service in a communication process according to Embodiment 3 of the present invention. In the third embodiment, a service control function entity is a SIP Server, and the calling party is configured. The terminal is a SIP videophone terminal, and the media stream address is a URL. The third embodiment mainly includes the following steps:
步骤 301 : 主叫终端摘机, 将摘机事件上报给 SIP Server。  Step 301: The calling terminal picks up the phone and reports the off-hook event to the SIP server.
设定主叫终端与 SIP Server之间订阅了终端状态上报事件, 当主叫终端摘 机时, 主叫终端用户状态上报给 SIP Server, 即将摘机事件上报给 SIP Server。 在本实施例中,可以将上报的终端用户状态看作是呼叫请求,上报的信息包含 主叫终端标识和主叫终端的状态信息。  The caller terminal and the SIP server are configured to subscribe to the terminal status report event. When the calling terminal is off-hook, the status of the calling terminal is reported to the SIP server, and the off-hook event is reported to the SIP server. In this embodiment, the reported end user status can be regarded as a call request, and the reported information includes the calling terminal identifier and the status information of the calling terminal.
步骤 302: SIP Server从策略服务器处获得主叫终端对应的媒体流的 URL 地址。  Step 302: The SIP Server obtains the URL address of the media stream corresponding to the calling terminal from the policy server.
在本实施例中,由于主叫终端是 SIP可视电话终端,则当主叫终端摘机时, In this embodiment, since the calling terminal is a SIP videophone terminal, when the calling terminal picks up the phone,
SIP Server将为主叫终端查询需要播放的媒体流的 URL地址,以便主叫终端在 未进行视频通话时播放媒体流。 The SIP Server queries the calling terminal for the URL address of the media stream that needs to be played, so that the calling terminal plays the media stream when no video call is made.
策略服务器根据预先设定的方式获取主叫终端需要播放的媒体流的 URL 地址, 包括但不限于:  The policy server obtains the URL address of the media stream that the calling terminal needs to play according to a preset manner, including but not limited to:
主叫终端预先订阅了媒体流(视频短片或视频广告信息等), 并建立了主 叫终端标识与媒体流 URL的对应关系, SIP Server将主叫终端标识发送给策略 服务器, 策略服务器根据已建立的主叫终端标识与媒体流 URL的对应关系查 找出所述 URL, 并返回给 SIP Server。  The calling terminal subscribes in advance to the media stream (video clip or video advertisement information, etc.), and establishes a correspondence between the calling terminal identifier and the media stream URL, and the SIP server sends the calling terminal identifier to the policy server, and the policy server is established according to the The corresponding relationship between the calling terminal identifier and the media stream URL finds the URL and returns it to the SIP Server.
步骤 303: SIP Server将接收到的 URL发送给主叫终端, 则主叫终端播放 所述 URL对应的媒体流。  Step 303: The SIP server sends the received URL to the calling terminal, and the calling terminal plays the media stream corresponding to the URL.
主叫终端根据所述 URL从流媒体服务器处获取该 URL对应的媒体流,并 向用户播放该媒体流。  The calling terminal acquires the media stream corresponding to the URL from the streaming media server according to the URL, and plays the media stream to the user.
在本实施例三中, 只要终端与 SIP Server之间订阅了终端状态上报事件, 则只要终端摘机且未进行视频通话时,都可以通过终端向用户播放设定的音视 频数据。  In the third embodiment, as long as the terminal and the SIP server subscribe to the terminal status report event, the set audio and video data can be played to the user through the terminal as long as the terminal goes off-hook and the video call is not performed.
如图 4所示, 为本发明实施例四中在通信过程中的信令流程示意图,在本 实施例四中, 设定主叫终端为 PSTN终端, 被叫终端为 SIP可视电话终端, 当 主叫终端在听回铃音且被叫终端在振铃时,被叫终端播放媒体流数据。此过程 包括: 步骤 401 : 主叫终端通过 SIP Server发起呼叫请求。 As shown in FIG. 4, it is a schematic diagram of a signaling flow in a communication process according to Embodiment 4 of the present invention. In the fourth embodiment, a calling terminal is a PSTN terminal, and a called terminal is a SIP videophone terminal. When the calling terminal listens to the ring back tone and the called terminal is ringing, the called terminal plays the media stream data. This process includes: Step 401: The calling terminal initiates a call request through the SIP server.
主叫终端将呼叫请求发送给 SIP Server, 该呼叫请求中包含主叫终端标识 和被叫终端标识。  The calling terminal sends a call request to the SIP server, where the call request includes the calling terminal identifier and the called terminal identifier.
在本实施例中是以 SIP消息为例来说明实现彩视业务的, 因此, 所述呼叫 请求可以为邀请消息(SIP INVITE )。  In this embodiment, the SIP message is taken as an example to describe the implementation of the color video service. Therefore, the call request may be an invite message (SIP INVITE).
步骤 402: SIP Server确定被叫终端具有播放媒体流的能力。  Step 402: The SIP Server determines that the called terminal has the capability of playing the media stream.
SIP Server才 据接收到的呼叫请求中的主叫终端标识和被叫终端标识确定 主叫终端为 PSTN终端, 被叫终端为 SIP可视电话终端。  The SIP Server determines that the calling terminal is a PSTN terminal and the called terminal is a SIP videophone terminal according to the calling terminal identifier and the called terminal identifier in the received call request.
步骤 403: SIP Server从策略服务器处获取被叫终端对应的媒体流地址。 策略服务器根据被叫终端标识查找出订阅的媒体流的地址, 并返回给 SIP Step 403: The SIP server acquires a media stream address corresponding to the called terminal from the policy server. The policy server finds the address of the subscribed media stream according to the called terminal identifier, and returns it to the SIP.
Ssrvsr。 Ssrvsr.
步骤 404: SIP Server将所述呼叫请求发送到被叫终端, 将呼叫接续。 Step 404: The SIP Server sends the call request to the called terminal to connect the call.
SIP Server接收到返回的媒体流地址后, 将主叫终端和被叫终端之间的呼 叫接续: 向被叫终端发送呼叫请求, 并接收被叫终端返回的 SIP 180响应。 After receiving the returned media stream address, the SIP Server connects the call between the calling terminal and the called terminal: sends a call request to the called terminal, and receives the SIP 180 response returned by the called terminal.
此时, 主叫终端听到回铃音, 被叫终端开始振铃。  At this time, the calling terminal hears the ring back tone, and the called terminal starts ringing.
步骤 405: SIP Server将媒体流地址发送被叫终端, 被叫终端根据该媒体 流地址播放对应的媒体流。  Step 405: The SIP server sends the media stream address to the called terminal, and the called terminal plays the corresponding media stream according to the media stream address.
与实施例四类似地, 当主叫终端为 SIP可视电话终端, 被叫终端为 PSTN 终端时, 同样可以利用主叫终端播放预先设定的媒体流, 如图 5所示, 为本发 明实施例五中通信过程的信令流程示意图,与实施例四中不同的是: SIP Server 判断出主叫终端具有播放媒体流的能力 ,将从策略服务器中获取的媒体流地址 发送给主叫终端 ,由主叫终端根据媒体流地址从流媒体服务器处获取并播放媒 体流。  Similar to the fourth embodiment, when the calling terminal is a SIP videophone terminal and the called terminal is a PSTN terminal, the calling terminal can also play the preset media stream, as shown in FIG. 5, which is implemented by the present invention. The signaling flow diagram of the communication process in the fifth embodiment is different from the fourth embodiment in that: the SIP server determines that the calling terminal has the capability of playing the media stream, and sends the media stream address obtained from the policy server to the calling terminal. The media stream is obtained and played by the calling terminal from the streaming server according to the media stream address.
当主叫终端和被叫终端都具有播放媒体流的能力时(如都是 SIP可视电话 终端), 则可以按照本发明实施例六对应的图 6所示的信令流程示意图来实现 通信过程。 SIP Server确定主叫终端和被叫终端都是 SIP可视电话终端, 则从 策略服务器中分别查询这两个终端对应的媒体流的地址,主叫终端在听到回铃 音后,和被叫终端在振铃后,分别根据媒体流的地址从流媒体服务器中获取并 播放媒体流。 在实施例三、 四、五、 和六都是在主叫终端和被叫终端之间未通话时实现 彩视业务的过程,后续实施例七、八和九中是在主叫终端和被叫终端之间的通 话建立后实现彩视业务的过程。 When the calling terminal and the called terminal both have the capability of playing the media stream (for example, all of the SIP videophone terminals), the communication process can be implemented according to the signaling flow diagram shown in FIG. 6 corresponding to the sixth embodiment of the present invention. . The SIP server determines that both the calling terminal and the called terminal are SIP videophone terminals, and then query the address of the media stream corresponding to the two terminals from the policy server, and the calling terminal hears the ring back tone, and the called party After the terminal rings, the media stream is obtained and played from the streaming server according to the address of the media stream. In the third, fourth, fifth, and sixth embodiments, the process of implementing the color vision service when the call between the calling terminal and the called terminal is not in the call, and the subsequent embodiments in the seventh, eighth, and ninth are at the calling terminal and the called party. The process of implementing the color vision service after the call between the terminals is established.
如图 7所示, 为本发明实施例七中,在主叫终端和被叫终端之间建立通话 后实现彩视业务的过程, 在本实施例中, 设定主叫终端为 PSTN终端, 被叫终 端为 SIP可视电话终端, 包括:  As shown in FIG. 7, in the seventh embodiment of the present invention, a process of implementing a color vision service after a call is established between a calling terminal and a called terminal, in this embodiment, setting the calling terminal as a PSTN terminal, The terminal is a SIP videophone terminal, including:
步骤 701: 主叫终端向 SIP Server发送呼叫请求, SIP Server将该呼叫请求 接续到被叫终端, 建立主叫终端和被叫终端之间的通话连接。  Step 701: The calling terminal sends a call request to the SIP server, and the SIP server connects the call request to the called terminal to establish a call connection between the calling terminal and the called terminal.
所述呼叫请求中包含了主叫终端标识和被叫终端标识, SIP Server接收到 呼叫请求后, 根据被叫终端标识将呼叫接续到被叫终端: SIP Server将 SIP INVITE发送到被叫终端, 并接收被叫终端返回的 SIP 180响应消息, 此时主 叫终端听见回铃音 ,被叫终端振铃。被叫终端摘机时,向 SIP Server发送 SIP 200 响应消息, SIP Server将应答响应消息( SIP ACK )返回至被叫终端后, 主、 被叫终端之间开始通话。  The call request includes a calling terminal identifier and a called terminal identifier. After receiving the call request, the SIP Server connects the call to the called terminal according to the called terminal identifier: the SIP Server sends the SIP INVITE to the called terminal, and The SIP 180 response message returned by the called terminal is received. At this time, the calling terminal hears the ring back tone, and the called terminal rings. When the called terminal picks up the phone, it sends a SIP 200 response message to the SIP Server. After the SIP Server returns the response message (SIP ACK) to the called terminal, the call between the primary and the called terminal starts.
步骤 702: SIP Server从策略服务器中获取被叫终端对应的媒体流的地址。 Step 702: The SIP server acquires an address of the media stream corresponding to the called terminal from the policy server.
SIP Server根据被叫终端标识确定被叫终端为 SIP可视电话终端, 则由策 略服务器查询出为被叫终端定制的媒体流的地址。 The SIP server determines that the called terminal is a SIP videophone terminal according to the called terminal identifier, and the policy server queries the address of the media stream customized for the called terminal.
步骤 703: 被叫终端接收到媒体流地址后, 播放该媒体流地址对应的媒体 流。  Step 703: After receiving the media stream address, the called terminal plays the media stream corresponding to the media stream address.
实施例八是以主叫终端为 SIP可视电话、 被叫终端为 PSTN终端为例的, 与实施例七的基本过程类似,区别在于:在主叫终端和被叫终端建立连接之后, SIP Server从策略服务器处查询出为主叫终端订阅的媒体流的地址, 由主叫终 端根据所述媒体流的地址从流媒体服务器处获取并播放对应的媒体流, 如图 8 所示。  In the eighth embodiment, the calling terminal is a SIP videophone and the called terminal is a PSTN terminal, which is similar to the basic process of the seventh embodiment. The difference is that after the calling terminal and the called terminal establish a connection, the SIP server The address of the media stream subscribed to the calling terminal is queried from the policy server, and the calling terminal obtains and plays the corresponding media stream from the streaming media server according to the address of the media stream, as shown in FIG. 8.
在实施例九中, 设定主叫终端和被叫终端都为 SIP可视电话终端, 本实 施例可以看作实施例七和实施例八的结合,如图 9所示,在主被叫终端建立呼 叫连接之后 , SIP Server从策略服务器处分别获取为主叫终端和被叫终端定制 的媒体流的地址,则主叫终端和被叫终端分别根据媒体流的地址从流媒体服务 器处获取并播放媒体流。 在本发明实施例中,可以统计主被叫终端播放媒体流的时长, 并将该时长 信息返回给用户。统计时长的方式包括但不限于: 在终端开始播放媒体流时启 动计时器, 在停止播放媒体流时停止计时器, 根据计时器获取播放时长; 具有 播放媒体流能力的终端与 SIP Server之间订阅终端状态上报事件, 当终端开始 或停止播放媒体流时可以通过 SIP等消息将播放状态发送给 SIP Server, SIP Server将该播放状态传递给 AS, AS根据启动播放事件和停止播放事件统计出 播放时长。 In the ninth embodiment, the calling terminal and the called terminal are both SIP videophone terminals. This embodiment can be regarded as a combination of Embodiment 7 and Embodiment 8, as shown in FIG. After the call connection is established, the SIP server obtains the addresses of the media streams customized by the calling terminal and the called terminal respectively from the policy server, and the calling terminal and the called terminal respectively obtain and play from the streaming media server according to the address of the media stream. Media stream. In the embodiment of the present invention, the duration of playing the media stream by the calling and called terminals may be counted, and the duration information is returned to the user. The method of counting the duration includes, but is not limited to, starting a timer when the terminal starts playing the media stream, stopping the timer when the media stream is stopped, and obtaining the playing duration according to the timer; and subscribing between the terminal having the capability of playing the media stream and the SIP Server The terminal status reports the event. When the terminal starts or stops playing the media stream, the broadcast status can be sent to the SIP server through a message such as SIP. The SIP server transmits the play status to the AS. The AS counts the play duration according to the start play event and the stop play event. .
在实施例三至实施例九中, 主、 被叫终端都是属于同一个 SIP Server, 在 本发明实施例中, 主、 被叫终端也可以分别属于不同的 SIP Server。 如图 10 所示, 为本发明实施例十中信令流程示意图, 主叫终端和被叫终端分别属于不 同的 SIP Server。在图中 1001标识的虚线框中表示被叫终端在振铃时播放媒体 流的过程; 图中 1002标识的虚线框中表示主叫终端在听回铃音时播放媒体流 的过程; 图中 1003标识的虚线框表示主叫终端和被叫终端进行非视频通话时 播放媒体流的过程。 在本实施例的图 10中, 是应用服务器将被叫终端订阅的 媒体流的地址返回给为被叫终端提供服务的 SIP Server, 也可以由为主叫终端 提供服务的 SIP Server转发该地址。  In the third embodiment to the ninth embodiment, the master and the called terminal belong to the same SIP server. In the embodiment of the present invention, the master and the called terminal may belong to different SIP servers. As shown in FIG. 10, it is a schematic diagram of a signaling process in the tenth embodiment of the present invention, where the calling terminal and the called terminal belong to different SIP servers. The dotted line box identified by 1001 in the figure indicates the process in which the called terminal plays the media stream when ringing; the dotted line box identified by 1002 in the figure indicates the process in which the calling terminal plays the media stream while listening to the ring back tone; The dashed box of the logo indicates the process of playing the media stream when the calling terminal and the called terminal perform a non-video call. In FIG. 10 of the embodiment, the application server returns the address of the media stream subscribed by the called terminal to the SIP server serving the called terminal, and may also forward the address by the SIP server serving the calling terminal.
实施例三至实施例六的彩视业务的实现是在主被叫终端之间未建立呼叫 连接时,实施例七至实施例十的彩视业务的实现是在主被叫终端之间建立非视 频呼叫连接后, 在本发明实施例中, 包括但不限于: 将呼叫连接建立之前的过 程和呼叫连接建立之后的过程结合在一起实现彩视业务。 如图 11所示, 为本 发明实施例十一的信令流程示意图, 在本实施例中, 主、 被叫终端都为 SIP可 视电话终端, 业务控制功能实体为归属域服务器节点(S-CSCF ), 在通话连接 之前, 分别播放订阅的媒体流, 在进行非视频通话之后, 主、 被叫终端仍然播 放订阅的媒体流。 该步骤包括:  The implementation of the color-vision service of the third embodiment to the sixth embodiment is implemented when the call connection is not established between the calling and the called terminals. The implementation of the color-vision service of the seventh embodiment to the tenth embodiment is to establish a non-connection between the calling and called terminals. After the video call is connected, in the embodiment of the present invention, including but not limited to: the process before the call connection is established and the process after the call connection is established to implement the color vision service. As shown in FIG. 11, FIG. 11 is a schematic diagram of a signaling flow according to Embodiment 11 of the present invention. In this embodiment, both the primary and the called terminals are SIP videophone terminals, and the service control function entity is a home domain server node (S- CSCF), before the call connection, respectively play the subscribed media stream, after the non-video call, the main and called terminals still play the subscribed media stream. This step includes:
步骤 1101 : 主叫终端通过 S-CSCF发起呼叫请求,该呼叫请求中可以包含 主、 被叫终端的标识。  Step 1101: The calling terminal initiates a call request through the S-CSCF, where the call request may include identifiers of the calling and called terminals.
当主叫终端摘机后即可发起所述呼叫请求,该呼叫请求可以是 INVITE消 息, 通过代理节点 (P-CSCF )发送到 S-CSCF, S-CSCF确定主被叫终端具有 播放媒体流的能力, 则将呼叫请求转发到应用服务器。 步骤 1102: S-CSCF从应用服务器处获得主、 被叫终端分别对应的媒体流 地址。 The call request may be initiated after the calling terminal picks up the phone, and the call request may be an INVITE message, which is sent to the S-CSCF through the proxy node (P-CSCF), and the S-CSCF determines that the calling and called terminals have the playing media stream. Capabilities, then forward the call request to the application server. Step 1102: The S-CSCF obtains, from the application server, a media stream address corresponding to the primary and the called terminal respectively.
应用服务器根据主、被叫终端标识查询出订阅的媒体流的地址, 并将该地 址通过 INVITE返回给 S-CSCF。  The application server queries the address of the subscribed media stream according to the primary and called terminal identifiers, and returns the address to the S-CSCF through the INVITE.
步骤 1103: S-CSCF将所述呼叫请求发送到被叫终端, 将呼叫接续。  Step 1103: The S-CSCF sends the call request to the called terminal to connect the call.
为主叫终端提供服务的 S-CSCF 将呼叫请求通过为被叫终端提供服务的 S-CSCF转发到被叫终端, 被叫侧返回 180响应消息后, 主叫终端处于听回铃 音状态, 被叫用户处于振铃状态。  The S-CSCF that provides the service to the calling terminal forwards the call request to the called terminal through the S-CSCF that provides the service to the called terminal. After the called side returns the 180 response message, the calling terminal is in the state of listening to the ring back tone. The user is called to be ringing.
步骤 1104: S-CSCF将媒体流地址发送到主、 被叫终端侧, 主、 被叫终端 获取所述地址对应的媒体流, 并播放。  Step 1104: The S-CSCF sends the media stream address to the calling terminal and the called terminal side, and the primary and the called terminal acquires the media stream corresponding to the address, and plays the media stream.
为主叫终端提供服务的 S-CSCF将从应用服务器处获得的媒体流地址分别 通过 P-CSCF和为被叫终端提供服务的 S-CSCF发送给主叫终端和被叫终端, 主、被叫终端根据所述地址从流媒体服务器处获取对应的媒体流, 并向用户播 放。  The S-CSCF serving the calling terminal sends the media stream address obtained from the application server to the calling terminal and the called terminal through the P-CSCF and the S-CSCF serving the called terminal respectively, the master and the called party. The terminal acquires the corresponding media stream from the streaming media server according to the address, and plays the content to the user.
在本实施例中, 也可以由应用服务器将所述地址发送给主、 被叫终端。 步骤 1105: 被叫终端摘机, 通话连接建立。  In this embodiment, the address may also be sent by the application server to the calling and called terminals. Step 1105: The called terminal picks up the phone, and the call connection is established.
在此步骤中, 应用服务器可以根据主、被叫终端的标识再次查询主、被叫 终端订阅的媒体流地址(主、被叫终端可以在不同的时间订阅不同的媒体流)。  In this step, the application server can query the media stream addresses subscribed by the calling and called terminals again according to the identifiers of the calling and called terminals (the primary and the called terminals can subscribe to different media streams at different times).
步骤 1106: 应用服务器将所述地址发送给主、 被叫终端, 主、 被叫终端 根据所述地址获得对应的媒体流, 并播放。  Step 1106: The application server sends the address to the calling terminal and the called terminal, and the primary and the called terminal obtain the corresponding media stream according to the address, and play the same.
若在步骤 1105中, 应用服务器未再次查询媒体流地址, 则此步骤中根据 步骤 1104中获得的地址播放媒体流。  If the application server does not query the media stream address again in step 1105, then the media stream is played in accordance with the address obtained in step 1104 in this step.
在实施例二至实施例十一中, 都是以 SIP协议作为数据传输的协议, 且具 有播放媒体流能力的终端为 SIP可视电话终端, 在本发明实施例十二中,设定 具有播放媒体流能力的终端为融合终端, 主叫终端和被叫终端都为融合终端, 通信流程如图 12 ( a )和图 12 ( b )所示, 图 12 ( a )表示主叫终端和被叫终 端之间未建立通话连接时实现彩视业务流程, 包括:  In the second embodiment to the eleventh embodiment, the SIP protocol is used as the data transmission protocol, and the terminal having the capability of playing the media stream is a SIP videophone terminal. In the twelfth embodiment of the present invention, the setting has a play. The terminal of the media stream capability is a converged terminal, and both the calling terminal and the called terminal are converged terminals, and the communication process is shown in FIG. 12(a) and FIG. 12(b), and FIG. 12(a) shows the calling terminal and the called party. The color vision service process is implemented when the call connection is not established between the terminals, including:
步骤 1201:主叫终端中的 PSTN终端通过 SIP Server向被叫终端中的 PSTN 终端发起呼叫请求, 该呼叫请求中包含主叫终端标识和被叫终端标识。 主叫终端中的 PSTN通过正常方式向被叫终端中的 PSTN发起请求。 Step 1201: The PSTN terminal in the calling terminal initiates a call request to the PSTN terminal in the called terminal by using the SIP server, where the call request includes the calling terminal identifier and the called terminal identifier. The PSTN in the calling terminal initiates a request to the PSTN in the called terminal in the normal manner.
主叫终端中的 PSTN终端将该呼叫请求通过初始地址消息( IAM )发送至 为主叫终端提供服务的 SIP Server A, SIP Server A将该呼叫请求通过 INVITE 消息转发至为被叫终端提供服务的 SIP Server B, SIP Server B将该呼叫请求再 次通过 IAM发送被叫终端的 PSTN终端。 被叫终端中的 PSTN终端将应答响 应消息原路返回至主叫终端中的 PSTN终端。  The PSTN terminal in the calling terminal sends the call request to the SIP Server A that provides the service to the calling terminal through the initial address message (IMM), and the SIP Server A forwards the call request to the called terminal through the INVITE message. SIP Server B, SIP Server B sends the call request again through the IAM to the PSTN terminal of the called terminal. The PSTN terminal in the called terminal returns the response response message to the PSTN terminal in the calling terminal.
步骤 1202: SIP Server确定主被叫终端具有播放媒体流的能力从策略服务 器处查询主被叫终端定制的媒体流的地址。  Step 1202: The SIP Server determines that the calling and called terminals have the ability to play the media stream, and query the address of the media stream customized by the calling and called terminals from the policy server.
SIP Server从策略服务器处获取媒体流的地址可以通过信息( INFO )消息 携带需要的信息(如被叫终端标识、 主叫终端标识和呼叫信息等)。  The SIP Server obtains the address of the media stream from the policy server and can carry the required information (such as the called terminal identifier, the calling terminal identifier, and the call information, etc.) through the information (INFO) message.
步骤 1203: 策略服务器将查询出的媒体流地址分别返回给为主、 被终端 提供服务的 SIP Server。  Step 1203: The policy server returns the queried media stream address to the primary SIP server that is served by the terminal.
在本实施例中, 可以通过 200响应消息携带返回的媒体流地址。  In this embodiment, the returned media stream address may be carried by the 200 response message.
步骤 1204: SIP Server分别将媒体流地址返回给主、被叫终端中 SIP终端。 设定主叫终端对应的媒体流地址为 URL A1 , 被叫终端对应的媒体流地址 为 URL B1 , 则 SIP Server利用 INFO消息将 URL A1返回给主叫终端中的 SIP 终端 , 将 URL B 1返回给被叫终端中 SIP终端。  Step 1204: The SIP server returns the media stream address to the SIP terminal in the calling and called terminals respectively. The media stream address corresponding to the calling terminal is set to URL A1, and the media stream address corresponding to the called terminal is URL B1, and the SIP server returns the URL A1 to the SIP terminal in the calling terminal by using the INFO message, and returns the URL B1. Give the SIP terminal in the called terminal.
步骤 1205: 主、 被叫终端分别接受到媒体流地址获取并播放媒体流。 主、 被终端向为自身提供服务的 SIP Server返回 200响应消息。  Step 1205: The primary and the called terminal respectively receive the media stream address to acquire and play the media stream. The primary and secondary terminals return a 200 response message to the SIP Server that provides services for itself.
通过步骤 1201至 1205,实现了在主、被叫终端建立通话之前的彩视业务。 如图 12 ( b )所示, 为主、 被叫终端之间建立通话后的彩视业务, 包括:  Through steps 1201 to 1205, the color picture service before the call is established between the primary and the called terminal is implemented. As shown in Figure 12 (b), the color vision service after the call is established between the primary and the called terminal, including:
步骤 1206: 主、 被叫终端之间建立正常通话连接。  Step 1206: Establish a normal call connection between the calling terminal and the called terminal.
被叫终端摘机, 将信息通过应答消息 (ANM )发送给为被叫终端提供服 务器的 SIP Server, 该 SIP Server将该信息通过 200消息转发至为主叫终端提 供服务器的 SIP Server, 该 SIP Server将该信息通过 ANM消息发送给主叫终 端的 PSTN终端, 并将 ACK消息返回给为被叫终端提供服务的 SIP Server。  The called terminal picks up the phone, and sends the information to the SIP server that provides the server for the called terminal through the response message (ANM), and the SIP server forwards the information to the SIP server of the server provided by the calling terminal through the 200 message, the SIP Server. The information is sent to the PSTN terminal of the calling terminal through the ANM message, and the ACK message is returned to the SIP Server serving the called terminal.
步骤 1207至步骤 1210与前述步骤 1202至 1205类似。  Steps 1207 through 1210 are similar to steps 1202 through 1205 described above.
步骤 1211 :主叫终端的 SIP终端将携带播放时长的 INFO消息通过为自身 提供服务的 SIP Server转发至策略服务器。 被叫终端返回播放时长的过程类似,被叫终端的 SIP终端也将携带播放时 长的 INFO消息通过为自身提供服务的 SIP Server转发至策略服务器。 Step 1211: The SIP terminal of the calling terminal forwards the INFO message carrying the playing duration to the policy server through the SIP server serving the self. The process in which the called terminal returns the playing duration is similar. The SIP terminal of the called terminal also forwards the INFO message carrying the playing duration to the policy server through the SIP server that provides the service itself.
策略服务器通过相同的路径返回 200响应消息。  The Policy Server returns a 200 response message over the same path.
根据实施例二至实施例十二的方案可知:需要主叫终端和被叫终端中的一 个或两个具有播放媒体流的能力, 对于终端本身的类型并不限制, 可以是 SIP 终端、 H.323终端或融合终端或分离终端, 因此, 通信过程可以是不同类型终 端之间的通信, 不限于前述实施例描述的情况。  According to the solution of the second embodiment to the twelfth embodiment, one or two of the calling terminal and the called terminal are required to have the capability of playing the media stream, and the type of the terminal itself is not limited, and may be a SIP terminal, H. 323 terminal or converged terminal or separate terminal, therefore, the communication process may be communication between different types of terminals, and is not limited to the case described in the foregoing embodiment.
在 SIP服务器向策略服务器要求查询媒体流地址时,信息可以通过 INFO、 MESSAGE等消息发送; 策略服务器将获取的媒体流的地址通过 200、 INFO, MESSAGE等消息返回给 SIP服务器。  When the SIP server requests the policy server to query the media stream address, the information can be sent through INFO, MESSAGE, etc.; the policy server returns the obtained media stream address to the SIP server through the 200, INFO, MESSAGE and other messages.
根据前面的系统及方法的描述,本发明实施例十三还提供一种业务控制功 能实体, 如图 13所示, 该业务控制功能实体包括接收模块 21、 终端能力确定 模块 22和获取模块 23 , 其中, 接收模块 21用于接收终端发送的呼叫请求, 该呼叫请求中包含终端标识; 终端能力确定模块 22用于确定所述终端标识对 应的终端是否具有播放媒体流的能力; 获取模块 23用于当所述终端标识对应 的终端具有播放媒体流能力时 , 获取所述终端订阅的媒体流的地址。  According to the description of the foregoing system and method, the thirteenth embodiment of the present invention further provides a service control function entity. As shown in FIG. 13, the service control function entity includes a receiving module 21, a terminal capability determining module 22, and an obtaining module 23. The receiving module 21 is configured to receive a call request sent by the terminal, where the call request includes a terminal identifier, and the terminal capability determining module 22 is configured to determine whether the terminal corresponding to the terminal identifier has the capability of playing a media stream. When the terminal corresponding to the terminal identifier has the capability of playing the media stream, the address of the media stream subscribed by the terminal is obtained.
所述业务控制功能实体还包括发送模块 24, 用于将所述媒体流的地址返 回给具有播放媒体流能力的所述终端。 在本实施例中, 发送模块 24可以将所 述媒体流地址携带在 INFO消息中发送给终端。  The service control function entity further includes a sending module 24, configured to return an address of the media stream to the terminal having the capability of playing a media stream. In this embodiment, the sending module 24 may carry the media stream address in an INFO message and send it to the terminal.
在本发明实施例所述描述的系统、 方法和业务控制功能实体中, 主、被叫 终端与 SIP Server之间需要实现信令传输, 若在 SIP协议中, 则以 SIP消息为 载体传输需要的信息, 若在 H.323协议中, 则以 H.323消息为载体传输需要的 信息。 以 SIP协议为例, SIP消息一般支持 TEL URI/SIP URI, 而由于本实施 例中, SIP消息需要携带媒体流服务器中媒体流的 RTSP ( Real-Time Streaming Protocol, 实时流协议) URL或 HTTP ( Hypertext Transfer Protocol, 超文本传 输协议) URL, 因此需要扩展 SIP 消息中的统一资源标识(URI )地址, 使 SIP消息也支持 RTSP URL或 HTTP URL。 接口信息如表 1所示: INFO sip: *889@10.75.35.161; user=phone SIP/2.0 In the system, the method, and the service control function entity described in the embodiments of the present invention, signaling transmission between the primary and the called terminal and the SIP server is required, and in the SIP protocol, the SIP message is used as a carrier for transmission. Information, if in the H.323 protocol, transmits the required information using the H.323 message as the carrier. For example, the SIP message generally supports the TEL URI/SIP URI. In this embodiment, the SIP message needs to carry the RTSP (Real-Time Streaming Protocol) URL or HTTP (for the media stream in the media stream server). Hypertext Transfer Protocol, Hypertext Transfer Protocol) URL, so you need to extend the Uniform Resource Identifier (URI) address in SIP messages so that SIP messages also support RTSP URLs or HTTP URLs. The interface information is shown in Table 1: INFO sip: *889@10.75.35.161; user=phone SIP/2.0
Via: SIP/2.0/UDP 10.70.106.82:5060;branch=z9hG4bK3883054875  Via: SIP/2.0/UDP 10.70.106.82:5060;branch=z9hG4bK3883054875
From: <sip: From: <sip:
075588881234@10.75.35.161;user=phone>;tag=1451640803 075588881234@10.75.35.161;user=phone>;tag=1451640803
To: <sip: 075589834250@10.75.35.161;user=phone>  To: <sip: 075589834250@10.75.35.161;user=phone>
Call-ID: EC78236491@10.70.106.82  Call-ID: EC78236491@10.70.106.82
CSeq: 29 INFO  CSeq: 29 INFO
Max-Forwards: 5  Max-Forwards: 5
Subject: Client Request  Subject: Client Request
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER,
SUBSCRIBE, NOTIFY, INFO, INFO SUBSCRIBE, NOTIFY, INFO, INFO
Content-Type: text/xml  Content-Type: text/xml
Content-Length: ..  Content-Length: ..
<inf> <inf>
<cmd>xxxx</ cmd>
Figure imgf000015_0001
<cmd>xxxx</ cmd>
Figure imgf000015_0001
<autolink>xxx</ autolink>  <autolink>xxx</ autolink>
</inf>  </inf>
表 1  Table 1
其中 , INFO消息为传送的消息格式, cmd命令字表示需要执行的动作, 如: cmd为 4001表示在主、 被叫终端在为进行视频通话时, 具有播放媒体 流能力的终端播放订阅的媒体流。  The INFO message is the format of the transmitted message, and the cmd command word indicates the action to be performed. For example, the cmd is 4001, indicating that the terminal having the media stream capability plays the subscribed media stream when the calling terminal and the called terminal are in the process of making a video call. .
<linkuri>表示媒体流的地址 , 地址类型可以是 HTTP URL或者 RTSP URL, 当地址类型为 HTTP URL时, 终端链接到对应的网站上; 地址类型 为 RTSP URL时, 终端链接到对应的流媒体服务器。  <linkuri> indicates the address of the media stream. The address type can be an HTTP URL or an RTSP URL. When the address type is an HTTP URL, the terminal is linked to the corresponding website. When the address type is RTSP URL, the terminal is linked to the corresponding streaming server. .
<cont>表示 linkuri链接的附力说明。 <event>表示终端启动、 停止、 暂停播放媒体流的状态事件。 <cont> indicates the attached description of the linkuri link. <event> indicates a status event in which the terminal starts, stops, and pauses the playback of the media stream.
<autolink>表示终端连接方式, 可以手动方式也可以自动方式。 如: autolink为 2时表示终端在收到消息后终端自动连接到流媒体服务器, 终端 不需要任何操作。  <autolink> indicates the terminal connection method, either manually or automatically. For example, if autolink is 2, the terminal automatically connects to the streaming server after receiving the message, and the terminal does not need any operation.
同样 SIP Server与应用服务器之间也需要实现信令传输,若在 SIP协议中 , 则以 SIP消息为载体传输需要的信息, 若在 H.323协议中, 则以 H.323消息为 载体传输需要的信息。 以 SIP协议为例, SIP消息一般支持 TEL URI/SIP URI, 而由于本实施例中, SIP消息需要携带媒体流服务器中媒体流的 RTSP URL或 HTTP URL , 因此需要扩展 SIP消息中的 URI地址, 使 SIP消息也支持 RTSP Similarly, the SIP server and the application server also need to implement signaling transmission. If the SIP protocol is used, the SIP message is used as the carrier to transmit the required information. If the H.323 protocol is used, the H.323 message is used as the carrier transmission. Information. For example, the SIP message generally supports the TEL URI/SIP URI. In this embodiment, the SIP message needs to carry the RTSP URL or the HTTP URL of the media stream in the media stream server, so the URI address in the SIP message needs to be extended. Make SIP messages also support RTSP
URL或 HTTP URL。 接口信息如表 2所示: URL or HTTP URL. The interface information is shown in Table 2:
INFO sip: 075589834250@10.75.35.161;user=phone SIP/2.0  INFO sip: 075589834250@10.75.35.161;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.70.106.82:5060;branch=z9hG4bK3883054875  Via: SIP/2.0/UDP 10.70.106.82:5060;branch=z9hG4bK3883054875
From: <sip: From: <sip:
075588881234@10.75.35.161;user=phone>;tag=1451640803 075588881234@10.75.35.161;user=phone>;tag=1451640803
To: <sip: 075589834250@10.75.35.161;user=phone>  To: <sip: 075589834250@10.75.35.161;user=phone>
Call-ID: EC78236491@10.70.106.82  Call-ID: EC78236491@10.70.106.82
CSeq: 29 INFO  CSeq: 29 INFO
Max-Forwards: 5  Max-Forwards: 5
Subject: Client Request  Subject: Client Request
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, INFO, INFO  Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, INFO, INFO
Content-Type: text/xml  Content-Type: text/xml
Content-Length: ..  Content-Length: ..
<inf> <inf>
<cmd>xxxx</ cmd>  <cmd>xxxx</ cmd>
"^^linl u i^^* **************** "^^/l lril un^^"
Figure imgf000016_0001
<autolink>xxx</ autolink>
"^^linl ui^^* **************** "^^/l lril un^^"
Figure imgf000016_0001
<autolink>xxx</ autolink>
■^^callee^^* ****************** *"^^/callee^^  ■^^callee^^* ****************** *"^^/callee^^
■^^callei*^^* ****************** *"^^/calle ^^ ■^^callei*^^* ****************** *"^^/calle ^^
Figure imgf000017_0001
Figure imgf000017_0001
</inf>  </inf>
表 2  Table 2
表 1和表 2是 INFO消息的接口信息, 如表 3所示, 为 200消息的接口信息<Table 1 and Table 2 are the interface information of the INFO message, as shown in Table 3, the interface information of the 200 message <
SIP/2.0 200 OK SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.70.106.82:5060;branch=z9hG4bK3883054875 Via: SIP/2.0/UDP 10.70.106.82:5060;branch=z9hG4bK3883054875
From: <sip:From: <sip:
075588881234@10.75.35.161;user=phone>;tag=1451640803 075588881234@10.75.35.161;user=phone>;tag=1451640803
To: <sip: 075589834250@10.75.35.161;user=phone>;tag=1330799 To: <sip: 075589834250@10.75.35.161;user=phone>;tag=1330799
Call-ID: EC78236491@10.70.106.82 Call-ID: EC78236491@10.70.106.82
CSeq: 29 INFO  CSeq: 29 INFO
Content-Type: text/xml  Content-Type: text/xml
Content-Length: ..  Content-Length: ..
<inf> <inf>
<cmd>xxxx</ cmd>  <cmd>xxxx</ cmd>
"^^linkuri^^"* **************** "^^/l lnkun^^"
Figure imgf000017_0002
"^^linkuri^^"* **************** "^^/l lnkun^^"
Figure imgf000017_0002
<autolink>xxx</ autolink>  <autolink>xxx</ autolink>
■^^callee^^* ****************** *"^^/callee^^  ■^^callee^^* ****************** *"^^/callee^^
■^^callei*^^* ******************* "^^/calle ^^" ■^^callei*^^* ******************* "^^/calle ^^"
Figure imgf000017_0003
Figure imgf000017_0003
</inf>  </inf>
表 3  table 3
其中, cmd、 linkuri、 cont、 event, autolink Tag参数同软交换和终端接口 含义。 Among them, cmd, linkuri, cont, event, autolink Tag parameters are the same as softswitch and terminal interface. Meaning.
<callee>表示被叫终端标识, <caller>表示主叫终端标识; <callinfo>表示正 常呼叫时信息。  <callee> indicates the called terminal identity, <caller> indicates the calling terminal identity; <callinfo> indicates normal call time information.
通过本发明所述的方法、 系统和业务控制功能实体在主、被叫终端未进行 视频通话时,播放预先订阅的媒体流数据,使得用户同时可以获得更多的信息, 体验较好。 明的精神和范围。这样,倘若本发明的这些修改和变型属于本发明权利要求及 其等同技术的范围之内 , 则本发明也意图包含这些改动和变型在内。  The method, system and service control function entity of the present invention play pre-subscribed media stream data when the main and called terminals do not make a video call, so that the user can obtain more information at the same time, and the experience is better. The spirit and scope of the Ming. Thus, it is intended that the present invention cover the modifications and variations of the inventions

Claims

权 利 要 求 Rights request
1、 一种通信方法, 其特征在于, 该方法包括:  A communication method, characterized in that the method comprises:
接收来自主叫终端的呼叫请求, 根据所述呼叫请求中的主叫终端标识和 / 或被叫终端标识确定具有播放媒体流能力的终端,获取该终端订阅的媒体流的 地址; 将所述媒体流的地址发送至所述具有播放媒体流能力的终端。  Receiving a call request from the calling terminal, determining, according to the calling terminal identifier and/or the called terminal identifier in the call request, a terminal having a capability of playing the media stream, and acquiring an address of the media stream subscribed by the terminal; The address of the stream is sent to the terminal having the capability to play the media stream.
2、 如权利要求 1所述的方法, 其特征在于, 所述方法还包括: 所述具有 播放媒体流能力的终端根据所述媒体流的地址获取并播放订阅的媒体流。  The method according to claim 1, wherein the method further comprises: the terminal having the capability of playing the media stream acquiring and playing the subscribed media stream according to the address of the media stream.
3、 如权利要求 1所述的方法, 其特征在于, 所述获取该终端订阅的媒体 流的地址的步骤包括:由业务控制功能实体从应用服务器处获取具有播放媒体 流能力的终端订阅的媒体流的地址。  The method according to claim 1, wherein the step of acquiring an address of the media stream subscribed by the terminal comprises: obtaining, by the service control function entity, the media subscribed by the terminal having the capability of playing the media stream from the application server The address of the stream.
4、 如权利要求 3所述的方法, 其特征在于, 所述主被叫终端同属一个业 务控制功能实体, 或分别属于不同的业务控制功能实体。  4. The method according to claim 3, wherein the calling and called terminals belong to the same one of the service control function entities, or belong to different service control function entities.
5、 如权利要求 4所述的方法, 其特征在于, 当所述主被叫终端分别属于 不同的业务控制功能实体, 且所述主被叫终端都具有播放媒体流能力时,  The method according to claim 4, wherein when the calling and called terminals respectively belong to different service control function entities, and the calling and called terminals have the capability of playing media streams,
所述将所述媒体流的地址发送至所述具有播放媒体流能力的终端包括: 所述应用服务器将主被叫终端订阅的媒体流的地址分别发送给主被叫终 端所属的业务控制功能实体,由所述主被叫终端分别所属的业务控制功能实体 发送至主被叫终端;  The sending the address of the media stream to the terminal having the capability of playing the media stream includes: the application server separately sending the address of the media stream subscribed by the calling and called terminals to the service control function entity to which the calling and called terminals belong Transmitting to the calling and called terminal by the service control function entity to which the calling and called terminals respectively belong;
或者,  Or,
所述应用服务器将主被叫终端订阅的媒体流的地址都发送给主叫终端所 属的业务控制功能实体,由所述主叫终端所属的业务控制功能实体将主叫终端 订阅的媒体流的地址发送给主叫终端 ,将被叫终端订阅的媒体流的地址发送给 被叫终端所属的业务控制功能实体,由所述被叫终端所属的业务控制功能实体 将被叫终端订阅的媒体流的地址发送给被叫终端。  The application server sends the address of the media stream subscribed by the calling terminal to the service control function entity to which the calling terminal belongs, and the address of the media stream subscribed by the calling terminal by the service control function entity to which the calling terminal belongs Sending to the calling terminal, sending the address of the media stream subscribed by the called terminal to the service control function entity to which the called terminal belongs, and the service control function entity to which the called terminal belongs to the address of the media stream subscribed by the called terminal Send to the called terminal.
6、 如权利要求 1所述的方法, 其特征在于, 所述方法还包括步骤: 所述业务控制功能实体将所述呼叫请求接续到被叫终端 ,  The method according to claim 1, wherein the method further comprises the step of: the service control function entity connecting the call request to the called terminal,
所述步骤应用于所述业务控制功能实体获取所述媒体流的地址之后; 或, 所述业务控制功能实体确定具有播放媒体流能力的终端之前。  The step is applied after the service control function entity acquires the address of the media stream; or the service control function entity determines before the terminal having the capability of playing the media stream.
7、 如权利要求 6所述的方法, 其特征在于, 如果被叫终端摘机通话连接 建立之前, 已获取终端订阅的媒体流的地址,将所述媒体流的地址发送至所述 具有播放媒体流能力的终端, 则在被叫终端摘机通话连接建立之后还可以包 括: 7. The method according to claim 6, wherein if the called terminal picks up the call connection Before the establishment, the address of the media stream subscribed by the terminal is obtained, and the address of the media stream is sent to the terminal having the capability of playing the media stream.
再次获取终端订阅的媒体流的地址,将所述媒体流的地址发送至所述具有 播放媒体流能力的终端。  The address of the media stream subscribed by the terminal is obtained again, and the address of the media stream is sent to the terminal having the capability of playing the media stream.
8、 如权利要求 2所述的方法, 其特征在于, 所述方法还包括:  The method of claim 2, wherein the method further comprises:
在具有播放媒体流的能力的所述终端开始播放媒体流时启动计时器,在停 止播放媒体流时停止计时器, ^居计时器获取播放时长; 或  The timer is started when the terminal having the capability of playing the media stream starts playing the media stream, the timer is stopped when the media stream is stopped, and the playing time is obtained by the timer; or
具有播放媒体流的能力的所述终端启动或停止播放媒体流时将播放状态 通过业务控制功能实体发送给应用服务器,该应用服务器根据启动播放事件和 停止播放事件统计出播放时长。  When the terminal having the capability of playing the media stream starts or stops playing the media stream, the playing state is sent to the application server through the service control function entity, and the application server counts the playing duration according to the start playing event and the stop playing event.
9、 如权利要求 1所述的方法, 其特征在于, 所述方法应用于会话发起协 议 SIP或 H.323协议;  9. The method of claim 1, wherein the method is applied to a Session Initiation Protocol SIP or H.323 protocol;
当应用会话发起协议时, 通过扩展 SIP消息中的统一资源标识 URI地址, 使 SIP消息支持实时流协议 RTSP URL或超文本传输协议 HTTP URL。  When the session initiation protocol is applied, the SIP message supports the real-time streaming protocol RTSP URL or the hypertext transfer protocol HTTP URL by extending the uniform resource identifier URI address in the SIP message.
10、 一种通信系统, 其特征在于, 该系统包括主叫终端、 被叫终端和业务 控制功能实体, 其中,  A communication system, comprising: a calling terminal, a called terminal, and a service control function entity, wherein
所述主叫终端, 用于向所述业务控制功能实体发起呼叫请求,该呼叫请求 中包含主叫终端标识和 /或被叫终端标识, 若具有播放媒体流的能力, 则根据 所述业务控制功能实体获取的媒体流的地址播放订阅的媒体流;  The calling terminal is configured to initiate a call request to the service control function entity, where the call request includes a calling terminal identifier and/or a called terminal identifier, and if the capability of playing the media stream is performed, according to the service control The address of the media stream obtained by the functional entity plays the subscribed media stream;
所述业务控制功能实体, 用于 ^居所述呼叫请求中的主叫终端标识和 /或 被叫终端标识确定具有播放媒体流能力的终端,并获取该终端订阅的媒体流的 地址; 将所述媒体流的地址发送至所述具有播放媒体流能力的终端;  The service control function entity is configured to determine, by the calling terminal identifier and/or the called terminal identifier in the call request, a terminal that has the capability of playing the media stream, and obtain an address of the media stream subscribed by the terminal; Sending an address of the media stream to the terminal having the capability of playing the media stream;
所述被叫终端, 用于若具有播放媒体流的能力, 则根据所述业务控制功能 实体获取的媒体流的地址播放订阅的媒体流。  The called terminal is configured to: if the device has the capability of playing the media stream, play the subscribed media stream according to the address of the media stream acquired by the service control function entity.
11、 如权利要求 10所述的系统, 其特征在于, 所述系统还包括: 应用服务器,用于查询出所述业务控制功能实体确定的具有播放媒体流能 力的终端订阅的媒体流的地址 , 并将该地址返回给所述业务控制功能实体。  The system of claim 10, wherein the system further comprises: an application server, configured to query, by the service control function entity, an address of a media stream subscribed by the terminal having the capability of playing the media stream, And returning the address to the service control function entity.
12、 如权利要求 10所述的系统, 其特征在于, 所述系统应用于会话发起 协议 SIP或 H.323协议。 12. The system of claim 10, wherein the system is applied to session initiation Protocol SIP or H.323 protocol.
13、 如权利要求 11所述的系统, 其特征在于, 所述系统还包括: 计时器,用于在具有播放媒体流的能力的所述终端开始播放媒体流时启动 计时器, 在停止播放媒体流时停止计时器, 统计播放时长; 或  The system according to claim 11, wherein the system further comprises: a timer, configured to start a timer when the terminal having the capability of playing the media stream starts playing the media stream, and stop playing the media Stop timer when streaming, count the duration of playback; or
具有播放媒体流的能力的所述终端,用于在启动或停止播放媒体流时将播 放状态发送给业务控制功能实体;  The terminal having the capability of playing a media stream, configured to send a play status to the service control function entity when starting or stopping playing the media stream;
所述业务控制功能实体, 用于将所述播放状态转发给应用服务器; 所述应用服务器, 用于根据启动播放事件和停止播放事件统计播放时长。 The service control function entity is configured to forward the play status to the application server, and the application server is configured to count the play duration according to the start play event and the stop play event.
14、 如权利要求 10所述的系统, 其特征在于, 所述具有播放媒体流的能 力的终端包括:会话发起协议可视电话终端、 H.323可视电话终端或融合终端。 The system according to claim 10, wherein the terminal having the capability of playing the media stream comprises: a session initiation protocol videophone terminal, an H.323 videophone terminal or a converged terminal.
15、 一种业务控制功能实体, 其特征在于, 该业务控制功能实体包括: 接收模块, 用于接收终端发送的呼叫请求, 该呼叫请求中包含终端标识; 终端能力确定模块,用于确定所述终端标识对应的终端是否具有播放媒体 流的能力;  A service control function entity, wherein the service control function entity comprises: a receiving module, configured to receive a call request sent by the terminal, where the call request includes a terminal identifier; and a terminal capability determining module, configured to determine the Whether the terminal corresponding to the terminal identifier has the capability of playing the media stream;
获取模块, 用于当所述终端标识对应的终端具有播放媒体流能力时,获取 所述终端订阅的媒体流的地址。  And an obtaining module, configured to acquire an address of the media stream subscribed by the terminal when the terminal corresponding to the terminal identifier has the capability of playing the media stream.
发送模块,用于将所述媒体流的地址返回给具有播放媒体流能力的所述终 端。  And a sending module, configured to return an address of the media stream to the terminal having the capability of playing a media stream.
16、 如权利要求 15所述的业务控制功能实体, 其特征在于, 所述发送模 块, 用于将携带所述媒体流地址的信息 INFO消息发送给所述终端。  The service control function entity according to claim 15, wherein the sending module is configured to send an information INFO message carrying the media stream address to the terminal.
17、 如权利要求 15所述的业务控制功能实体, 其特征在于, 所述业务控 制功能实体包括: 会话发起协议 SIP服务器或归属域服务器节点 S-CSCF。  The service control function entity according to claim 15, wherein the service control function entity comprises: a session initiation protocol SIP server or a home domain server node S-CSCF.
PCT/CN2008/072232 2007-09-13 2008-09-02 A communication method, system and service controlling function entity WO2009033401A1 (en)

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