TELECOMMUNICATION TERMINAL AND METHOD FOR CONTROLLING MEDIA STREAMS

Abstract

A terminal can have at least one audio interface for at least one microphone, at least one audio interface for at least one speaker, an operator control module and at least one interface for connecting to at least one telephony- and/or IP-telephony-compatible terminal. The terminal can be configured as a media hub for the at least one telephony- or IP-telephony-compatible terminal. The terminal can have a media processing unit for mixing and a media hub controller or controlling media streams received by the at least one telephony- or IP-telephony-compatible terminal and the at least one telephony- or IP-telephony-compatible terminal can be operated through the operator control module of the telecommunication terminal. A method for controlling media streams through a media hub controlled integrated into a telecommunication terminal is also disclosed.

Claims

1-10. (canceled)

11. A telecommunication terminal comprising: at least one audio interface for at least one microphone, at least one audio interface for at least one speaker, an operator control module, and at least one interface for connecting to at least one telephony- and/or IP-telephony-compatible terminal, the telecommunication terminal configured as a media hub for the at least one telephony- or IP-telephony-compatible terminal, wherein the telecommunication terminal has a media hub controller to control media streams received from the at least one telephony- or IP-telephony-compatible terminal and a media processing unit to mix the media streams, and wherein the operator control module on the telecommunication terminal is communicatively connectable to the at least one telephony- or IP-telephony-compatible terminal to operate the at least one telephony- or IP-telephony-compatible terminal.

12. The telecommunication terminal of claim 11, wherein the the media hub controller controls the media streams for output through the audio interface to the at least one speaker and/or for output to at least one telephony- or IP-telephony compatible terminal via the interface; and wherein the medias steams include audio streams.

13. The telecommunication terminal of claim 12, wherein the media hub controller is compatible with a local area network (LAN), a wireless LAN (W-LAN), a universal serial bus (USB), and/or Bluetooth.

14. The telecommunication terminal of claim 12, wherein the media processing unit is configured to mix media streams, and is also equipped with a transcriber function for converting the audio streams to text data.

15. The telecommunication terminal of claim 14, wherein the media hub controller is operable via the operator control module of the telecommunication terminal.

16. The telecommunication terminal of claim 15, wherein the media hub controller is configurable using the operator control module.

17. The telecommunication terminal of claim 16, wherein the telecommunication terminal is a VOID- compatible telephone, a SIP-compatible telephone, or a WebRTC-compatible telephone.

18. The telecommunication terminal of claim 11, wherein the interface is a universal serial bus (USB) and/or a Bluetooth and/or an Intranet and/or local area network (LAN) and/or wireless LAN (W-LAN) interface.

19. The telecommunication terminal of claim 11, including the at least one telephony-compatible or IP-telephony-compatible terminal, wherein the at least one telephony-compatible or IP-telephony-compatible terminal is a personal computer, a smart phone, a tablet device, and/or a laptop that is connectable to the telecommunication terminal through the interface, and wherein media hub extension software is installed on the telephony-compatible or IP-telephony-compatible terminal.

20. A method for controlling media streams through a media hub controller integrated into a telecommunication terminal, wherein the method comprises: connecting at least one telephony- or IP-telephony-compatible terminal with the telecommunication terminal through an interface; receiving media streams from the at least one telephony- or IP-telephony-compatible terminal at the telecommunication terminal, controlling the received media streams by the media hub controller to cause the media streams to be forwarded to at least one audio interface on the telecommunication terminal so that the media streams are output through at least one speaker attached to an audio interface of the telecommunication terminal, and controlling at least one telephony- or IP-telephony-compatible terminal through an operator control module on the telecommunication terminal.

Description

BRIEF DESCRIPTION OF THE DRAWINGS

[0019] The invention is described below in more detail based on preferred exemplary embodiments and with reference to the figures. The figures show:

[0020] FIG. 1 a schematic representation of a telecommunication terminal according to one embodiment of the invention;

[0021] FIG. 2 a flow chart of a telecommunication terminal according to one embodiment of the invention; and

[0022] FIG. 3 a sequence diagram for a call from a smart phone; and

[0023] FIG. 4 another sequence diagram for a WebRTC call and additional VoIP and smart phone calls.

REFERENCE NUMERALS USED IN THE DRAWINGS INCLUDE

[0024] 1 Telecommunication terminal/telephone

[0025] 2, 2 Smart phone

[0026] 3 Bluetooth connection

[0027] 4 Laptop

[0028] 5 USB connection

[0029] 6 PC

[0030] 7 Internet

[0031] 8 LAN/WLAN connection

[0032] 9 Media hub controller

[0033] 10 Telephone receiver

[0034] 11 Speaker

[0035] 12 Audio interface

[0036] 13 Audio interface

[0037] 14 Audio interface

[0038] 15 Audio interface

[0039] 16 Operator control module

[0040] 17 Screen

[0041] 18 Media processing unit

[0042] 19 Telephone unit

[0043] 20 Circuit controller unit

[0044] 21 Media sources

[0045] 22 LAN controller & driver

[0046] 23 USB controller & driver

[0047] 24 Bluetooth controller & driver

[0048] 25 Media sinks

[0049] 26 LAN controller and driver

[0050] 27 USB controller and driver

[0051] 28 Bluetooth controller and driver

[0052] 29 Media hub driver

[0053] 30 Sound system

[0054] 31 Cloud transcription service

[0055] 32 Media hub extension software

[0056] 33 Incoming call (smart phone)

[0057] 34 Ring tone

[0058] 35 Transcription session

[0059] 36 Media hub transcriber service

[0060] 37 Audio data (external audio)

[0061] 38 Call details (smart phone)

[0062] 39 Archive file

[0063] 40 Transcribed items

[0064] 41 Monitor

[0065] 42 Order to end

[0066] 43 Virtual sound driver on the PC

[0067] 44 WebRTC service

[0068] 45 WebRTC browser

[0069] 46 WebRTC controller

[0070] 47 SIP telephone/VoIP/PBX

[0071] 48 Headphones/Headset

[0072] 49 Microphone

DETAILED DESCRIPTION

[0073] FIG. 1 is a schematic representation of a telecommunication terminal 1 according to one embodiment of the invention. The terminal 1 includes hardware such as a processor connected to a non-transitory computer readable medium (e.g. memory, flash memory, etc.). The terminal 1 can also include at least one transceiver as well as other hardware (e.g. input devices such as a microphone etc., output devices such as a speaker or a display, etc.).

[0074] As can be seen here, various terminals, applications, or services running on terminals can be connected in a variety of ways to the telecommunication terminal 1, hereinafter designated as telephone 1. In the configuration shown here, a smart phone 2 is connected to the telephone 1 through a Bluetooth connection 3, a laptop 4 is connected to the telephone 1 through a USB connection 5, a PC 6 is connected to the telephone 1 through the Internet 7, with either a LAN connection or wireless LAN connection 8, and a Web RTC-based cloud service 44, here Circuit, is also connected or connectable to the telephone 1 through the Internet 7, with either a LAN connection or wireless LAN connection 8, in such a way that the media hub functions of the media hub controller 9, which is integrated into the telephone 1, are used.

[0075] The telephone 1 has a customary telephone receiver 10 with a built-in speaker and a built-in microphone, and a room speaker 11 and a room microphone 12 for open listening and hands-free talking without using the receiver 10. In addition, a headset or headphones 48 can be connected to the telephone 1, so that a speaker/microphone pair can be used as a third possibility.

[0076] With respect to the terminals connected to the telephone 1 here, the media hub or media hub controller 9 of the telephone 1 acts as a virtual sound system. Audio playback of audio streams received from the connected terminals takes place on the telephone 1either on the telephone receiver 10, which has the advantage that others are not bothered in a large office area, for exampleor on the headset 13 or room speaker 11. Which of the audio or speaker outputs 12, 13, 14, 15 is selected can be chosen or preset using the media hub controller 9 or entered through a corresponding operator control module 16 of the telephone 1 that controls the media hub controller 9 or can take place implicitly, for example by picking up the receiver 10. In the event that multiple playback processes are active in parallel, they can be mixed by a media processing unit integrated into the telephone 1 (see FIG. 2) and played in parallel.

[0077] It is also possible for a microphone in the telephone 1, such as the microphone 49, to be used as the speech input device. These recorded audio streams then flow to all connected terminals 2, 4, 6 and can be used there for recording, for telephoning (e.g., via the smart phone 2 or the cloud service 44), or for other media hub additional functions such as recording or transcribing on the PC 6 or laptop 4.

[0078] If a smart phone 2 connected to the telephone 1 via the Bluetooth connection 3 is used, the telephone 1 can direct calls received through the smart phone 2. The Bluetooth hands-free profile is used for this, i.e., hanging up, placing, or accepting calls via the smart phone 2 can be done on the telephone 1.

[0079] If a WebRTC-based cloud service 44, such as Circuit, is used on the PC 6, laptop 4, or smart phone 2, the telephone 1 also acts as an amplifier/speaker/microphone unit by means of the virtual sound system function mentioned above. Neither an additional PC headset nor a speaker/microphone unit is needed for this. Calls received through the cloud service (e.g., Circuit) 44, for example, can be signaled like normal VoIP calls to the telephone 1 and can also be directly received, forwarded, and ended there.

[0080] If an active conversation exists, such as a WebRTC call through Circuit, additional incoming calls on other terminals, such as the smart phone 2 or telephone 1 itself, can be signaled as busy. If an incoming call is accepted, in actuality a device-overlapping conference session is generated and the data are appropriately mixed on the media processing unit (see FIG. 2) and respectively forwarded via the media hub controller 9.

[0081] However, the telephone 1 itself can also be connected to Circuitwithout a PC 6, laptop 4, or similar deviceand thereby make or accept Circuit RTC calls. This has the advantage that it is not necessary to first initiate a Circuit session on the PC 6 or laptop 4, because that device is not yet switched on, for example, and therefore no other Circuit calls are missed. If the user now logs on in parallel to Circuit from a laptop 4 or PC 6 using his access data, he can control the running Circuit-RTC session either from here or from the telephone 1, muting it or hanging up, for example.

[0082] In addition, additional Media Hub software can be installed on the PC 6, laptop 4, or smart phone 2 that enables additional functions. For example, a recording function can be used there and can be activated or stopped easily as needed on the telephone 1 through the operator control module 16, which is appropriately expanded for that purpose. Recordings are filed in a media hub archive for later use. In addition to audio data, the telephone 1 also delivers voice connection details, including connected subscriber names/numbers and times, etc. An add-on feature of the media hub software can allow transcription, i.e. conversion of audio streams to text data, on the PC 6, laptop 4, or smart phone 2. It converts the spoken audio stream into a text format using external transcription service providers (e.g., Nuance NDEV or Google Cloud Speech). As with recording, this can easily be started and stopped using the telephone 1 by means of the operator control module 16 attached to it and expanded to allow that function. The resulting text can be saved and archived together with the recording in a corresponding file format. Examples of this would be MP4 or MKV.

[0083] Text transcribed in this manner can be very helpful in facilitating searches for information. It is also an advantage for hearing-impaired people, for example, if the transcribed text of a telephone conversation is displayed at the same time on a screen, such as a screen 17 on the PC 6, so that the hearing-impaired person can more easily follow the conversation by reading the displayed text.

[0084] Obviously it is also possible for all terminals attached or connected to the telephone 1 to use this service. For example, a call on a smart phone 2, which is connected to the media hub controller 9 and the media processing unit 18 (see FIG. 2) of the telephone 1 through a Bluetooth connection 3, can be recorded or transcribed on the PC 6 or laptop 4 that is also connected. In this case, the audio streams from the smart phone 2 would be forwarded through the telephone 1 to the PC 6 or laptop 4 and then recorded there. The recording that is made can be stored in the memory of the PC 6 or laptop 4, for example.

[0085] Alternatively, if an audio playback process is started on the PC 6 that is connected to the telephone 1, such as playing an audio book, and starts the media/audio hub additional functions on the smart phone 2 as described above, then transcription would take place and be displayed there if applicable.

[0086] FIG. 2 is a flow diagram of a telecommunication terminal or telephone 1 according to one embodiment of the invention. In the configuration shown here, the sequence is controlled entirely by the expanded operator control module 16. Using the operator control module 16, the user can control various functions of the telephone 1 acting as media hub or media bridge, e.g., each of the terminals 2, 4, 6, which can function as media sources 21 or media sinks 25, can be switched actively or passively, for example by means of graphic operation of a touch screen for the operator control module 16. The operator control module 16 includes keys for that, and a screen to display functions and also, for example, for operating the telephone 1, if the screen is equipped as a touch screen, for example. The functions controlled through the operator control module 16 include Start and Stop for audio recording and text transcription. It is also possible, through the operator control module, to control or set whether audio signals are blocked or whether they should be mixed in the media processing unit 18, which can be the case, for example, if two or more media sources 21 are active in parallel.

[0087] In this case the telephone 1 actually also functions as a type of conference bridge between different attached devices, such as the PC 4 or laptop 6, a smart phone 2 or 2, etc., wherein the integrated telephone function of the telephone unit and the Circuit Client function of the Circuit controller unit 20 can also be included as connected terminals.

[0088] The media hub of the telephone 1 consists of a media hub controller 9 and a media processing unit 18. The media hub controller 9 handles all the logic and controls the media streams, such as audio streams or video streams, for example. It is also in contact logically with the connected terminals 4, 6, 2, and 2 and sends commands in their direction or receives orders and results from them. The Circuit controller unit 20 controls the connection to the Circuit system. The media hub controller 9 is configured or controlledas already described aboveby the expanded operator control module 16. Audio streams as well as commands and results from/to the attached terminals 2, 2, 4, and 6 are sent and received through the LAN/USB/Bluetooth controller (LAN controller and driver 22, USB controller and driver 23, Bluetooth controller and driver 24), including the respective applicable drivers.

[0089] The media sinks designated as 25 in the drawing references are also included in the telephone 1.

[0090] The PC 6 or laptop 4 connected to the telephone 1 is also equipped with a LAN/USB/Bluetooth controller (LAN controller and driver 26, USB controller and driver 27, Bluetooth controller and driver 28), a media hub driver 29, a sound system 30, and media hub extension software 32 for recording and transcription.

[0091] FIG. 3 is a sequence diagram for a call from a smart phone 2, which is controlled by Media Hub and is transcribed and archived using media hub extension software 32 on a connected PC workstation 6.

[0092] In this case, the smart phone 2 first executes pairing with the telephone 1 and therefore through the Bluetooth connection 3 with the media hub controller 9 that is integrated into the telephone 1. The media hub controller 9 then executes pairing with the PC 6 and therefore through the LAN connection 8 with the media hub extension software 32 that is installed on the PC 6.

[0093] As soon as a new incoming call 33 is signaled by the smart phone 2, the media hub controller 9 announces the call by means of a ring tone 34 that is output from the speaker 14 on the telephone 1. At the same time, the incoming call 33 is signaled to the expanded operator control module 16, after which the incoming call is accepted through the expanded operator control module 16 on the telephone 1, by lifting the receiver 10, for example (see FIG. 1). After the call is accepted, audio data from the smart phone 2 is directed to the media hub controller 9 and from the media hub controller 9 to the smart phone 2.

[0094] Then the media hub controller 9 starts a transcription session 35 on the PC 6 through the media hub extension software 32 installed on it. This initiates a media hub transcriber service 36, which access a cloud transcription service 31. The audio data 37 and call details 38 from the smart phone 2, received by the media hub controller 9, are forwarded from there to the media hub extension software 32 on the PC 6, where an archive file 39 for the audio data and call details is created by the media hub extension software 32. The data placed in the archive file are then forwarded to the cloud transcription service 31, which transcribes the audio data, i.e., sends converted and transcribed items 1, 2, . . . N 40 to the PC 6, where the transcribed items 1, 2, . . . N are displayed on a monitor 41 on the PC 6 and are archived by the media hub transcriber service 36.

[0095] This process ends when the call 42 is ended through the expanded operator control module 16 on the telephone 1, by hanging up the receiver 10, for example (see FIG. 1), which ends the call coming from the smart phone 2. At the same time, the media hub controller 9 orders the media hub transcriber service 36 on the PC 6 to end the transcription session, which causes the media hub transcriber service 36 to also order the cloud transcription service 31 to end the session.

[0096] FIG. 4 is another sequence diagram for a WebRTC call from a cloud service 44 (cloud WebRTC service) and additional VoIP and smart phone calls. It shows how a Circuit RTC call is handled by the telephone 1 and how additional SIP or smart phone calls can be refused as busy or alternatively can also be accepted and mixed in the telephone 1 by the media processing unit 17.

[0097] Here also, the smart phone 2 is paired through a Bluetooth connection 3 with the telephone 1 by its media hub controller 9, which causes the telephone 1 to be paired by its media hub controller 9 with the PC 6 through its media hub extension software 32. Next, a virtual sound driver 43 on the PC 6 is registered on the media hub extension software 32. Through a cloud-based WebRTC service 44, an incoming call on a WebRTC browser 45 on the PC 6 is accepted at the PC 6, and the media hub extension software 32 is notified of the incoming and accepted call. The media hub extension software 32 then notifies the media hub controller 9 on the telephone 1 about the accepted WebRTC call, and the media hub controller 9 notifies a WebRTC controller 46 or Circuit/WebRTC controller 46 about the accepted call. The details of the call are then displayed on a screen, which can be a component of the expanded operator control module 16 on the telephone 1, for example. At the same time, a busy signal is output from the telephone 1.

[0098] The WebRTC browser 45 of the PC 6 then forwards recorded audio data (sound playback) of the WebRTC call to the virtual sound driver 43, from which these data are forwarded to the media hub extension software 32 and from there to the media hub controller 9 on the telephone 1, where they are then output on the speaker 14. Audio streams are also received on the microphone 12 of the telephone 1 and forwarded to the media hub controller 9, which then forwards them to the media hub extension software 32 on the PC 6, from which they are sent to the virtual sound driver 43 and from there to the microphone output on the WebRTC browser 45.

[0099] During this process, i.e., during the active call, in which the telephone 1 and the PC 6 are participating through the WebRTC service 44, another call comes from the SIP telephone 47 (it can also be a VoIP telephone or a PBX), an (SIP INVITE), which is forwarded to the media hub controller 9. The media hub controller then queries the WebRTC controller 46 for the status, causing it to send a busy signal. The media hub controller 9, which receives the busy signal, correspondingly forwards it (486-BUSY) to the SIP telephone 47.

[0100] The same procedure takes place if a call comes in to the smart phone 2; it is signaled to the media hub controller 9, which again queries the WebRTC controller 46 for the status and receives a busy signal, which it signals as busy to the smart phone 2 in order to refuse the incoming call on the smart phone 2.

[0101] The WebRTC call can be ended by the user via the operator control module 16 on the telephone, for example by pressing a key or hanging up the receiver 10. This informs the media hub controller 9 that the call should be ended. It sends the instruction to end the call to the WebRTC controller 46, which sends a corresponding instruction to the media hub extension software 32 on the PC 6 and notifies the expanded operator control module 16 on the telephone 1, which displays on its integrated screen that the call has been ended. The media hub extension software 32 on the PC sends an instruction to end the WebRTC call to the WebRTC browser 45, which forwards it to the WebRTC service 44 so that the call is then definitively ended on all devices involved.

[0102] While certain present preferred embodiments of a communication apparatus, communication system, communication device, communication terminal, non-transitory computer readable medium, and embodiments of methods for making and using the same have been shown and described above, it is to be distinctly understood that the invention is not limited thereto but may be otherwise variously embodied and practiced within the scope of the following claims.