WO2012175964A2 - Procédés et systèmes de téléconférence multi-partie - Google Patents

Procédés et systèmes de téléconférence multi-partie Download PDF

Info

Publication number
WO2012175964A2
WO2012175964A2 PCT/GB2012/051433 GB2012051433W WO2012175964A2 WO 2012175964 A2 WO2012175964 A2 WO 2012175964A2 GB 2012051433 W GB2012051433 W GB 2012051433W WO 2012175964 A2 WO2012175964 A2 WO 2012175964A2
Authority
WO
WIPO (PCT)
Prior art keywords
audio data
call
telephony device
party
received
Prior art date
Legal status (The legal status 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 status listed.)
Ceased
Application number
PCT/GB2012/051433
Other languages
English (en)
Other versions
WO2012175964A3 (fr
Inventor
Neil James COLLINS
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Metaswitch Networks Ltd
Original Assignee
Metaswitch Networks 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 Metaswitch Networks Ltd filed Critical Metaswitch Networks Ltd
Publication of WO2012175964A2 publication Critical patent/WO2012175964A2/fr
Publication of WO2012175964A3 publication Critical patent/WO2012175964A3/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

Links

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00—Data switching networks
    • H04L12/02—Details
    • H04L12/16—Arrangements for providing special services to substations
    • H04L12/18—Arrangements for providing special services to substations for broadcast or conference, e.g. multicast
    • H04L12/1813—Arrangements for providing special services to substations for broadcast or conference, e.g. multicast for computer conferences, e.g. chat rooms
    • H04L12/1822—Conducting the conference, e.g. admission, detection, selection or grouping of participants, correlating users to one or more conference sessions, prioritising transmission
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/40—Support for services or applications
    • H04L65/403—Arrangements for multi-party communication, e.g. for conferences
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/20—Automatic or semi-automatic exchanges with means for interrupting existing connections; with means for breaking-in on conversations
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/38—Graded-service arrangements, i.e. some subscribers prevented from establishing certain connections
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • H04M3/568—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities audio processing specific to telephonic conferencing, e.g. spatial distribution, mixing of participants
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M2203/00—Aspects of automatic or semi-automatic exchanges
    • H04M2203/50—Aspects of automatic or semi-automatic exchanges related to audio conference
    • H04M2203/5009—Adding a party to an existing conference

Definitions

  • the present disclosure relates to multi-party teleconference methods and systems.
  • Multi-party audio teleconferences today are a vital business tool for allowing people to "meet" without being in the same location.
  • a multi-party audio teleconference typically comprises a number (greater than two) of endpoint telephony devices, either wire line or wireless, and an audio teleconference bridge.
  • the endpoint telephony devices dial into the multi-party audio teleconference bridge, enter an access code and can then talk to other people in the multi-party audio teleconference call.
  • the way this tends to work is that the multi-party audio teleconference bridge detects several of the loudest (or "active") speakers, mixes the audio from these speakers and sends out the mixed audio to all participants but subtracts the audio that each individual is sending to the audio teleconference bridge to avoid echo or feedback.
  • a method of handling an incoming telephone call of a first call type during a multi-party teleconference call which is of a second call type, different from the first call type comprising:
  • embodiments provide for handling a call, such as a particular, new type of multi-party teleconference call, which is interruptible, differently to a standard call type, for example.
  • the multi-party teleconference call is interruptible such that when a given participant in the multi-party teleconference call receives an incoming telephone call originating from outside of the multiparty teleconference call, the participant is removed from the multi-party teleconference call and the incoming telephone call is directed to the participant's telephony device.
  • Embodiments therefore avoid the telephony devices of multi-party teleconference call participants from being engaged and hence unavailable for incoming calls.
  • the handling comprises disabling the first call leg between the first telephony device and the multi-party teleconference service.
  • the disabling comprises tearing down the first call leg between the first telephony device and the multi-party teleconference service.
  • network resources in and out of multi-party teleconference service can be released and used for other purposes.
  • the disabling comprises maintaining the network resources reserved for establishing the first call leg, but not transmitting any audio data from the first telephony device to the multi-party teleconference service. Hence, the time taken in reserving network resources when re-entering the multi-party teleconference call is reduced.
  • the handling comprises directing the incoming telephone call setup request to the first telephony device, whereby the first telephony device enters a ringing state in relation to the incoming telephone call.
  • the user is notified of the incoming call by a ringing tone emitting from their telephony device.
  • the user need take no action to exit the call multi-party teleconference call, i.e. no user input is required for the incoming call to be directed to their telephony device.
  • Embodiments comprise, in response to a user of the first telephony device answering the incoming call, establishing a second call leg for transmittal of audio data associated with the answered telephone call between the first telephony device and the further telephony device. Hence, the user can choose to take the incoming call as normal.
  • Other embodiments comprise, in response to termination of the answered telephone call, tearing down the second call leg between the first telephony device and the further telephony device; and establishing a third call leg for transmittal of audio data associated with the multi-party teleconference call between the first telephony device and the multi-party teleconference service. Hence, once the incoming call has finished, the user can be reconnected back into the multi-party teleconference call.
  • the establishing of the third call leg comprises applying modified signalling to the third call leg such that the third call leg will be automatically answered by the first telephony device. Hence, no user input, other than hanging up the incoming call is required from the user in order to reenter them into the multi-party teleconference call.
  • the establishing of the third call leg comprises instructing the first telephony device to operate in speakerphone mode.
  • a user when a user enters the multi-party teleconference call, they will automatically hear any audio produced by all other multi-party teleconference call participants without having to pick up their telephony device handset. Also, any audio produced by a user will be transmitted to all other multi-party teleconference call participants without the user having to speak directly into the handset of their telephony device.
  • the first telephony device is capable of operating either in handset mode or speakerphone mode, during the multi-party teleconference call, the first telephony device operates in speakerphone mode, and the handling comprises instructing the first telephony device to operate in handset mode.
  • Embodiments comprise receiving an incoming multi-party teleconference call setup request from the first telephony device, the incoming multi-party teleconference call setup request comprising an identifier associated with the multi-party teleconference call, and the multi-party teleconference call in respect of which the first call leg is established is identified on the basis of the received identifier.
  • the user will be entered into the correct multi-party teleconference call, for example by recognition of a telephone dialling number associated with the first telephony device.
  • establishing the third call leg comprises utilising the maintained network resources. Hence, re-entry back into the multi-party teleconference call can be expedited.
  • Some embodiments comprise, in response to a user of the first telephony device not answering the first telephony device in the ringing state for a predetermined time period, aborting the handling and re-enabling the first call leg between the first telephony device and the multi-party teleconference service. Hence, if a user chooses not to or is unable to take the incoming call, they will re-enter the multi-party teleconference call.
  • a first telephony device of said at least three telephony devices, a first mixed audio data stream generated using a first audio mixing mode, in which said first mixed audio data stream is generated from a plurality of received audio data streams, including a first received audio data stream and a second received audio data stream each having, in said first audio mixing mode, a respective volume adjustment level setting which results in a non-zero contribution to the first mixed audio data stream, the volume adjustment level settings of said first and second received audio data streams having first respective levels in said first audio mixing mode;
  • embodiments provide for volume adjustment of the received data streams in order to provide improved functionality upon the occurrence of certain trigger events.
  • the first respective volume adjustment levels comprise substantially equal volume adjustment levels, whereby each audio data stream in the plurality has a substantially equal volume adjustment level in the first mixed audio data stream.
  • the second respective levels comprise a relatively high volume adjustment level for the first received audio data stream and a relatively low volume adjustment level for the second received audio data stream, in the second audio mixing mode.
  • the second respective levels may comprise a plurality of substantially equal volume adjustment levels, including the adjustment level for the second received audio data stream.
  • the second respective levels comprise a plurality of substantially equal volume adjustment levels, including the adjustment level for said first received audio data stream.
  • substantially equal volume adjustment levels of the first respective volume adjustment levels may receive a higher volume adjustment level than de-emphasized, substantially equal volume adjustment levels, of the second respective volume adjustment levels.
  • the volume adjustment level contribution to the mixed audio data streams due to for example background chatter can be lowered when a user raises their voice or two users are conducting a direct conversation via the multi-party teleconference.
  • Embodiments comprise transmitting the second mixed audio data stream generated using the second audio mixing mode to the first telephony device for a predetermined time period after detection of the trigger. Hence, a user's voice may be boosted, for example, over the duration of an announcement.
  • the trigger comprises a mean input volume of the first received audio data stream rising above a predetermined mean input volume.
  • Embodiments involve storing data representative of one or more predetermined words, and monitoring the received audio data streams for the utterance of any of the one or more predetermined words represented in the stored data, and the trigger comprises the monitoring detecting utterance of a given one or more predetermined words represented in the stored data.
  • embodiments can dynamically adjust the processing of audio data in the multiparty teleconference call to implement predetermined functionality.
  • the predetermined functionality can be triggered by one of the participants uttering one or more specific key words or phrases.
  • the given one or more predetermined words are uttered by a first user associated with a second telephony device of the at least three telephony devices, the given one or more predetermined words comprise an identifier for a second user associated with a third telephony device of the at least three telephony devices.
  • the given one or more predetermined words are uttered by a first user associated with a second telephony device of the at least three telephony devices
  • the given one or more words comprise an identifier for a second user associated with a third telephony device of the at least three telephony devices
  • the given one or more words further comprise an indication that the first user wishes to conduct a telephone call with the second user separate to the multi-party teleconference call
  • the second mixed audio data stream is generated in the second, different respective volume adjustment levels from audio data streams in the plurality apart from audio data stream received from the second and third telephony devices.
  • Embodiments comprise, in response to detecting the trigger during the multi-party teleconference call, transmitting the audio data stream received from the second telephony device to the third telephony device, transmitting the audio data stream received from the third telephony device to the second telephony device, and not transmitting the audio data streams received from the second and third telephony devices to the other participants.
  • two participants may have an exclusive conversation with each other, whilst the multi-party teleconference call may continue for the remaining other multi-party teleconference call participants.
  • a speech recognition function can be used to allow a participant to alter the mixing mode applied, for example by the utterance of a word or phrase.
  • apparatus adapted to perform the method of the first, second or third embodiments.
  • a computer program product comprising a non-transitory computer-readable storage medium having computer readable instructions stored thereon, the computer readable instructions being executable by a computerized device to cause the computerized device to perform the method of the first, second or third embodiments.
  • Figure 1 shows a system diagram according to embodiments.
  • Figure 2 shows a block diagram according to embodiments.
  • Figure 3 shows a flow diagram according to embodiments.
  • Figure 4 shows a graph according to embodiments.
  • Figure 5 shows a graph according to embodiments.
  • Figure 6 shows a graph according to embodiments.
  • Figure 1 shows a system diagram according to some embodiments.
  • Figure 1 depicts a telecommunications network 110 in which audio teleconferencing server (ATS) 100 is responsible for providing multi-party audio teleconference services to telephony devices TD1, TD2, TD3, TD4, TD5 and TD6.
  • ATS 100 may also be responsible for providing audio teleconference services to other telephony devices (not shown).
  • Each of TD1, TD2, TD3, TD4, TD5 and TD6 is associated with a user who works for or is otherwise associated with a given group or organisation.
  • Some of telephony devices TD1, TD2, TD3, TD4, TD5 and TD6 may be located in one office of the organisation, some may be located in one or more other offices of the organisation and some may be located at home locations where the respective users reside.
  • a user may be associated with more than one telephony device, for example a user may have one telephony device located on their desk at work and another telephony device located at their home.
  • Telephony device TD1 is provided with telephony services by telephone switch 104 which is connected to ATS 100 via network 102.
  • Network 102 may comprise one or more Public Switch Telephone Networks (PSTNs) and/or the Internet.
  • PSTNs Public Switch Telephone Networks
  • a user of telephony device TD1 also has access to a computing device 114 with display and user input capabilities, for example a personal computer, laptop, or personal digital assistant, etc. which is located locally to telephony device TD1.
  • a computing device 114 with display and user input capabilities for example a personal computer, laptop, or personal digital assistant, etc. which is located locally to telephony device TD1.
  • Telephone switch 104 is responsible for providing switching of telephone calls for a number of telephony devices such as telephony device TD1 including provision of dial tone, ringing tone, etc. Telephone switch 104 may also include the ability to select processes that can be applied to such calls, routeing for such calls based on signalling and subscriber database information, the ability to transfer control of calls to other network elements and management functions such as provisioning, fault detection and billing. Telephone switch 104 may also be referred to as a local telephone exchange, central office, class 5 switch or Softswitch. Telephone switch 104 includes a database 108 for storing call related data, including call state information, for example in relation to telephone calls incoming to or outgoing from telephony device TD1.
  • Telephony devices TD2, TD3, TD4, TD5 and TD6 may similarly be provided with telephony services by one or more other telephone switches (not shown).
  • telephony device TD7 is provided with telephony services by the same telephone switch 104 as telephony device TD1, but such services could be provided by a different telephone switch.
  • FIG. 2 shows a block diagram according to embodiments.
  • Figure 2 shows some components of ATS 100 of Figure 1.
  • ATS 100 comprises a processor 200 (or processors) for carrying out data processing functionality.
  • ATS 100 includes a data store 206 for storing data representative of predetermined words or phrases, and a data store 208 for storing user configuration data.
  • ATS 100 has an audio monitoring module 204 with speech recognition capabilities for monitoring audio data streams received from telephony devices TDl to TD6.
  • ATS 100 has a web interface 210 for providing user access to various multi-party audio teleconference functionalities such as display and user configuration.
  • ATS 100 has a mixing module 202 for processing audio data streams received from telephony devices TDl to TD6 including combining the received audio data streams into one or more mixed audio data streams according to different audio mixing modes.
  • Mixing module 202 includes codec functionality for decoding audio data streams received from telephony devices in different data formats and recoding mixed audio data streams for transmittal out to the telephony devices in the appropriate data formats.
  • Some embodiments relate to handling an incoming telephone call of a first call type during a multi-party audio teleconference call which is of a second call type, different from the first call type.
  • the second call type may for example be a call that can be interrupted.
  • the first call type may for example be a call that can not be interrupted.
  • ATS 100 provides a multi-party audio teleconference service and a multi-party audio teleconference call is currently being conducted between TD2, TD3, TD4, TD5 and TD6.
  • the user of TDl wishes to join the multi-party audio teleconference so dials an appropriate number associated with ATS 100 which results in an incoming multi-party audio teleconference call setup request being received at telephone switch 104.
  • Telephone switch 104 establishes a first call leg for transmittal of audio data associated with the multi-party audio teleconference call between TDl and ATS 100.
  • ATS 100 connects TDl to the audio teleconference call being conducted between TD2 to TD6.
  • Telephone switch 104 stores, in state information store 108, state information indicating that the call between TDl and TD2 to TD6 is of the second call type.
  • the second call type is an interruptible multi-party audio teleconference call type. This indicates that it can be interrupted by an incoming call to TDl. Alternatively, it may be indicative of a more general call type, for example an interruptible call type.
  • TD7 dials a telephone dialling number associated with TDl which results in an incoming call setup request from TD7 being received at telephone switch 104.
  • TD7 is not a participant in the multi-party audio teleconference call currently being conducted between TDl and TD2 to TD6.
  • Telephone switch 104 inspects the stored state information for the multi-party audio teleconference call and realises that the multi-party audio teleconference is of the second call type so can be interrupted with respect to TDl by the incoming call from TD7.
  • Telephone switch 104 therefore proceeds to handle the incoming telephone call setup request and the multi-party audio teleconference call in accordance with the stored state information indicating that the multiparty audio teleconference call is of the second call type.
  • Telephone switch 104 directs the incoming telephone call setup request to TDl which results in TDl ringing, i.e. TDl enters a ringing state in relation to the incoming telephone call from TD7.
  • Telephone switch 104 disables the first call leg between TDl and ATS 100.
  • the audio teleconference call between TD2 to TD6 carries on as normal, but without TDl participating.
  • telephone switch 104 establishes a second call leg for transmittal of audio data associated with the answered telephone call between TDl and TD7.
  • the users of TDl and TD7 are thus able to conduct a telephone call to each other.
  • ATS 100 continues to provide a multi-party audio teleconference service to TD2 to TD 6 and the audio teleconference call between TD2 to TD6 continues as normal.
  • telephone switch 104 When the user of TDl or the user of TD7 terminates the telephone call between them, for example by hanging up their respective telephony device, telephone switch 104 receives signalling information indicating such and tears down the second call leg between TDl and TD7. Telephone switch 104 establishes a third call leg for transmittal of audio data associated with the multiparty audio teleconference call between TDl and ATS 100. TDl thus re-enters the multi-party audio teleconference call being conducted between TD2 to TD6, i.e. the audio teleconference call then has all of TDl to TD7 participating again.
  • TDl is capable of operating either in a handset mode or a speakerphone mode.
  • Speakerphone mode is a mode of operation where a loudspeaker on TDl outputs audio from TDl without the user having to pick up the handset and put it to their ear/mouth.
  • an external (i.e. external to the handset) microphone on TDl picks up audio generated by the user (possibly also audio generated by others in close proximity and other background noise), without the user having to pick up the handset and put it to their ear/mouth.
  • TDl When the user of TDl initially joins the multi-party audio teleconference, the user dials an appropriate number associated with ATS 100 without picking up the handset of TDl . TDl thus enters the multi-party audio teleconference in speakerphone mode, i.e. during the multi-party audio teleconference call, TDl operates in speakerphone mode.
  • the appropriate number associated with ATS 100 may for example be stored in a speed-dial function on TD 1 such that the user of TD 1 just needs to press a single button to join the multi-party audio teleconference initially.
  • telephone switch 104 When telephone switch 104 handles the incoming call setup request received from TD7, telephone switch 104 instructs TDl to operate in handset mode. This means that TDl will cease to pick up audio from its external microphone. Further, no audio data from the disabled first call leg with ATS 100 will be output by the loudspeaker of TD 1. However, when telephone switch 104 directs the incoming telephone call setup request to TDl and it enters a ringing state, TDl is in handset mode and will emit a ringing sound from its loudspeaker.
  • TDl When the user of TDl picks up the handset of TDl to conduct the telephone call with the user of TD7, TDl will be in handset mode and continue to be in handset mode when the call with TD7 is terminated and the user of TDl puts the handset of TD1 back on-hook. However, when TD1 re-enters the multi-party audio teleconference call with TD2 to TD6, TD1 should be in speakerphone mode. Therefore, when telephone switch 104 establishes the third call leg between TD1 and ATS 100, telephone switch 104 instructs TD1 to operate in speakerphone mode.
  • telephone switch 104 applies modified signalling to the third call leg such that the third call leg will be automatically answered by TD 1.
  • the incoming multi-party audio teleconference call setup request comprises an identifier associated with the multi-party audio teleconference call being conducted between TD2 and TD7 which allows telephone switch 104 to recognise which multi-party audio teleconference call the incoming multi-party audio teleconference call setup request call relates to, i.e. one provided by ATS 100.
  • Telephone switch 104 therefore forwards the incoming multi-party audio teleconference call setup request on to ATS 100 and establishes the first call between TD1 and ATS 100 on the basis of the received identifier.
  • the received identifier comprises a telephone dialling number associated with TD1, for example detected from a calling line identifier (CLI) field of the incoming multi-party audio teleconference call setup request.
  • CLI calling line identifier
  • the received identifier comprises a telephone dialling number reserved for multi-party audio teleconference calls conducted via ATS 100, with telephone switch 104 being preconfigured to recognise multiparty audio teleconference call setup requests to the reserved dialling number and forward them to ATS 100 accordingly.
  • telephone switch 104 disables the first call leg between TDl and ATS 100 by tearing down the first call leg between TDl and ATS 100.
  • telephone switch 104 disables the first call leg between TDl and ATS 100 by maintaining the network resources reserved for establishing the first call leg, but not transmitting any audio data from TDl to ATS 100 for the duration of the incoming call from TD7.
  • telephone switch 104 establishes the third call leg by utilising the maintained network resources. This avoids having to establish the third call leg from scratch and can help speed up re-entry back into the multi-party audio teleconference call.
  • telephone switch 104 will abort the handling of the incoming call and re- enable the first call leg between TDl and ATS 100.
  • Figure 3 shows a flow diagram according to some embodiments.
  • Figure 3 depicts the flow of signalling messages and audio data between various entities of Figure 1 when a multi-party audio teleconference call is interrupted by an incoming call from a telephony device which is not participating in the multi-party audio teleconference call.
  • a multi-party audio teleconference call is currently being conducted between TDl, TD2, and TD3.
  • Other telephone devices may also participate in the multi-party audio teleconference call.
  • a call leg for transmittal of audio data associated with the multi-party audio teleconference call has been established between TD2 and ATS 100 as shown by step 3 a.
  • This call leg may be established via a telephone switch (not shown) which provides telephony services to TD2.
  • a call leg for transmittal of audio data associated with the multi-party audio teleconference call has also been established between TD3 and ATS 100 as shown by step 3b.
  • This call leg may be established via a telephone switch (not shown) which provides telephony services to TD3.
  • Telephone switch 104 establishes a call leg for transmittal of audio data associated with the multi-party audio teleconference call between TDl and ATS 100 as shown by steps 3c and 3d.
  • Telephone switch 104 stores state information indicating that the multi-party audio teleconference call between TDl to TD3 is of the second call type and is thus a multi-party audio teleconference which can be interrupted by an incoming call to TDl .
  • TD7 is not a participant in the multi-party audio teleconference call currently being conducted between TDl to TD3.
  • Telephone switch 104 inspects stored state information for the multi-party audio teleconference call and recognises that the multi-party audio teleconference is of the second call type and so can be interrupted with respect to TDl by the incoming call from TD7.
  • Telephone switch 104 therefore proceeds to handle the incoming telephone call setup request and the multi-party audio teleconference call in accordance with the stored state information indicating that the multi-party audio teleconference call is of the second call type.
  • Telephone switch 104 directs the incoming telephone call setup request to TDl in step 3g which results in TDl ringing, i.e. TDl enters a ringing state in relation to the incoming telephone call from TD7.
  • Step 3g also involves telephone switch 104 instructing TDl to operate in handset mode.
  • step 3f telephone switch 104 disables the first call leg between TDl and ATS 100.
  • the audio teleconference call between TD2 and TD3 carries on as normal, but without TDl participating.
  • step 3h the user of TDl chooses to answer the incoming call from TD7, such action being notified to telephone switch 104 in step 3i.
  • Telephone switch 104 establishes a second call leg for transmittal of audio data associated with the answered telephone call between TDl and TD7 in steps 3j and 3k.
  • the users of TD1 and TD7 are thus able to conduct a telephone call to each other.
  • ATS 100 continues to provide a multi-party audio teleconference service to TD2 and TD3 and the audio teleconference call between TD2 and TD3 carries on as normal.
  • step 31 the user of TD7 terminates the telephone call between TD1 and TD7, such action being notified to telephone switch 104 in step 3m.
  • Telephone switch 104 tears down the second call leg between TD1 and TD7.
  • Telephone 104 informs ATS 100 that TD1 is re-entering the multi-party audio teleconference call being conducted between TD2 and TD3 in step 3n.
  • telephone switch 104 instructs TD1 to operate in speakerphone mode.
  • Telephone switch 104 establishes a third call leg for transmittal of audio data associated with the multi-party audio teleconference call between TD1 and ATS 100 in steps 3p and 3q.
  • telephone switch 104 applies modified signalling to the third call leg such that the third call leg will be automatically answered by TD1, for example in conjunction with step 3o.
  • TD1 thus re-enters the multi-party audio teleconference call being conducted between TD2 and TD3, i.e. the audio teleconference call then has all three of TD1, TD2 and TD3 participating again.
  • Some embodiments relate to controlling a multi-party audio teleconference call in a telecommunications network. Such control is carried out by ATS 100 in relation to a multi-party audio teleconference call being conducted between at least three telephony devices in the network, for example telephony devices TD1, TD2, TD3, TD4, TD5 and TD6.
  • ATS 100 in relation to a multi-party audio teleconference call being conducted between at least three telephony devices in the network, for example telephony devices TD1, TD2, TD3, TD4, TD5 and TD6.
  • ATS 100 receives an audio data stream from each of telephony devices TD1, TD2, TD3, TD4, TD5 and TD6.
  • Processor 200 of ATS 100 processes the received audio data streams and mixing module 202 of ATS 100 combines the received audio data streams into mixed audio data streams.
  • Different mixed audio streams are then transmitted to the telephony devices TDl, TD2, TD3, TD4, TD5 and TD6 participating in the multi-party audio teleconference.
  • a mixed audio data stream transmitted to a given telephony device during the multi-party audio teleconference will not contain audio data received from that telephony device.
  • a mixed audio stream transmitted to TD6 will contain a mix of the audio data streams received from TDl to TD5, but no audio data from the audio data stream received from TD6.
  • a mixed audio data stream is generated from a plurality of received audio data streams using an audio mixing mode.
  • An audio mixing mode will generate a mixed audio data stream by combining the received audio data streams in the plurality according to respective volume adjustment levels defined for that audio mixing mode.
  • the mixing mode is normally set to a default audio mixing mode, in which each party provides a substantially equal contribution, and the volume adjustment levels for each incoming audio data stream included in any particular mixed audio data stream are equal.
  • the ATS 100 switches to a non-default audio mixing mode, in which the volume adjustment levels for the respective streams are altered.
  • the ATS 100 switches back to the default audio mixing mode.
  • a first mixed audio data stream is generated using a first audio mixing mode, for example the default audio mixing mode, and is transmitted to TD6.
  • the first mixed audio data stream is generated from a plurality of audio data streams in first respective volume adjustment levels, the plurality of data streams here being at least some of the audio data streams received from TDl to TD5.
  • a second mixed audio data stream is generated using a second audio mixing mode and is transmitted to TD6.
  • the second mixed audio data stream is generated from a plurality of audio data streams in second respective volume adjustment levels, the plurality of data streams here being at least some of the audio data streams received from TD1 to TD5.
  • the ATS 100 determines mean input volumes for each data stream.
  • the mean input volumes may be determined according to various known techniques, for example the mean input volume may be determined as an RMS (Root Mean Square) value of the incoming signal, to represent the power of the incoming signal.
  • the mean input volume of each of the received audio data streams is preferably measured over a periodic or sliding window of between 0.5 to 5 seconds, preferably between 1 and 2 seconds.
  • the ATS 100 determines volume adjustment levels for each data stream. These volume adjustments may be applied as a pre-amplification levels, applied in a pre-amplifier, before mixing of each of the audio data streams in equal proportions by an audio mixer, or alternatively may be applied as mixing levels, applied in differing proportions, in an audio mixer.
  • the mixing levels may represent fixed, linear, amplification levels or variable, non-linear, amplification levels.
  • the ATS 100 determines a normalised total output volume for each mixed data stream, and normalises the volume adjustment levels accordingly. This is shown in each of Figures 4 to 6, where the horizontal axis denotes time and the vertical axis denotes normalised total output volume of the mixed media data streams.
  • the normalised total output volume of a mixed media stream may be normalised against various parameters, for example a predetermined acceptable total output volume, or against a measurement of the input volumes of the incoming data streams. For example, it may be normalised against an average of the mean input volumes of each of the received audio data streams.
  • the normalised total output volume is normalised against an average of the mean input volumes of each of the received audio data streams, taken over a periodic or sliding window of between 0.5 to 5 seconds, preferably between 1 and 2 seconds.
  • the normalised total output volume is, in certain audio mixing modes, relatively low, for example in the region of 0.4 to 0.9, in this embodiment 0.6.
  • the normalised total output volume of the mixed media stream can be seen to increase to a relatively high level, e.g. in the region of 0.9 to 1.5, in this embodiment 1.0.
  • this provision of a normalised output volume in default mode which is below the normalised output volume in a non-default mode provides the advantage that, when a recipient of the mixed audio stream receives the non- default mode mixed stream, they do not need to turn down the output volume on their telephone speaker since this is normally set to an acceptable level comparable to the normalised, relatively high level, of the non-default mode mixed stream.
  • volume adjustment levels used for each of the received audio data streams Tl to T5 are shown schematically, to illustrate their respective sizes.
  • the first respective volume adjustment levels comprise substantially equal volume adjustment levels, whereby each input audio data stream in the plurality (i.e. TD1 to TD5) has a substantially equal volume adjustment level applied to generate the first mixed audio data stream transmitted to TD6.
  • the audio data streams received from TDl to TD5 make a non-zero, substantially equally adjusted, contribution, but there is no contribution from the audio data stream received from TD6.
  • a second mixed audio data stream generated using a second audio mixing mode is transmitted to TD6.
  • the second mixed audio data stream is generated from the same plurality of data streams, but the respective volume adjustment levels are different, the plurality of data streams again being the audio data streams received from TDl to TD5.
  • mixed audio data streams comprising different mixes of received audio data streams will also be transmitted to the other telephony devices TDl to TD5.
  • the second respective volume adjustment levels comprise a relatively high volume adjustment level applied to a first audio data stream received from TDl and relatively low, substantially equal, volume adjustment levels applied to the other audio data streams in the plurality, i.e. the audio data streams received from TD2 to TD5.
  • the volume adjustment level applied to TDl may for example be at least double that applied to each of TD2 to TD5.
  • each of the audio data streams received from TD2 to TD5 has a substantially equal volume adjustment, that there is a relatively higher contribution from the audio data stream received from TDl and that there is no contribution from the audio data stream received from TD6.
  • all the audio data streams received from TDl to TD5 make a non-zero contribution, but there is no contribution from the audio data stream received from TD6.
  • the trigger comprises the mean input volume of the first received audio data stream rising above a first predetermined threshold.
  • a first predetermined threshold This could for example be due to the user of TDl raising their voice to make an announcement to others in the room in which the user is located.
  • Increasing the contribution due to the voice of the user of TDl in the mixed audio data streams will enable the announcement to be more easily distinguished in the mixed audio data streams.
  • Increasing the contribution due to the voice of the user of TDl in the mixed audio data streams could continue for a predetermined time period, and/or after the detection of a predetermined period of silence (e.g. detected as a signal in which mean input volume is below a predetermined threshold) from the voice of the user of TDl, for example until time t2.
  • a predetermined period of silence e.g. detected as a signal in which mean input volume is below a predetermined threshold
  • the second respective volume adjustment levels comprise relatively high, substantially equal, volume adjustment levels applied to both a first audio data stream received from TD1 and a second audio data stream received from TD2 and relatively low, substantially equal, volume adjustment levels applied to the other audio data streams in the plurality, i.e. the audio data streams received from TD3 to TD5.
  • the volume adjustment levels applied to TD1 and TD2 may for example be at least double that applied to each of TD3 to TD5.
  • each of the audio data streams received from TD3 to TD5 has a substantially equal volume adjustment, that there is a relatively higher contribution from the audio data stream received from TD1 and TD2 and that there is no contribution from the audio data stream received from TD6.
  • all the audio data streams received from TD1 to TD5 make a non-zero contribution, but there is no contribution from the audio data stream received from TD6.
  • the trigger is detected at time tl and could comprise the mean input volume of one or more of the first received audio data stream and the second received audio data stream rising above a second predetermined threshold.
  • This could for example be due to the users of TDl and TD2 having a relatively loud conversation with each other.
  • Increasing the volume adjustment level of the voices of the users of TDl and TD2 in the mixed audio data streams will enable their conversation to be more easily distinguished in the mixed audio data streams thus helping them to converse more easily above contributions to the mixed audio data streams due to background noise or chatter.
  • Increasing the volume adjustment levels of the voice of the users of TDl and TD2 in the mixed audio data streams could continue for a predetermined time period after the detection, for example until time t2.
  • Embodiments comprise storing data representative of one or more predetermined words and monitoring the received audio data streams for the utterance of any of the one or more predetermined words represented in the stored data.
  • Monitoring module 204 of ATS 100 includes speech recognition capabilities, which may be embodied by any suitable speech recognition engine known in the art, such that when a user of a telephony device utters any of the one or more predetermined words, this can be detected in the audio data stream received from that user's telephony device.
  • the trigger comprises the monitoring detecting utterance of a given one or more predetermined words represented in the stored data in a received audio data stream.
  • the given one or more predetermined words are uttered by a user associated with TDl and are contained in the audio data stream received from TDl .
  • the given one or more predetermined words comprise an identifier for a user associated with TD2.
  • Such embodiments allow two users to conduct a conversation with each other within the multi-party audio teleconference. The volume adjustment level of the conversation between the two users is increased in the resulting mixed audio data stream which allows their conversation to be distinguished more easily.
  • the identifier could comprise the name of one of the two users such that the conversation can be initiated by one of the users uttering the name of the other user.
  • the given one or more predetermined words are uttered by a first user associated with TDl and the given one or more words comprise an identifier for a second user associated with TD2.
  • the given one or more words further comprise an indication that the first user wishes to conduct a private telephone call with the second user separate to the multi-party audio teleconference call.
  • the second mixed audio data stream is thus generated with second, different respective volume adjustment levels for the audio data streams in the plurality apart from audio data streams received from TDl and TD2.
  • the audio data stream received from TDl is transmitted to TD2 and the audio data stream received from the TD2 is transmitted to TDl , with substantially equal volume adjustment levels applied in each case - similar to a standard two-party telephone call.
  • the users of TD 1 and TD2 are thus able to have a private telephone call separate to the multi-party audio teleconference call and the multi-party audio teleconference call carries on between the remaining telephony devices (TD3 to TD5).
  • the indication that the first user wishes to conduct a telephone call with the second user separate to the multi-party audio teleconference call may comprise the first user uttering one or more key words, or a phrase, such as "private call" which are predetermined as being operable to trigger a private call, plus an identifier for the second user such as "Hey Joe".
  • each of the audio data streams received from TD3 to TD5 has a substantially equal volume adjustment level and contributes substantially equally to the total volume of the first mixed audio data stream, and that there is no contribution from any of the audio data streams received from TDl, TD2 or TD6.
  • Each of the audio data streams from TDl and TD2 are thus cut from the mixed audio data stream sent to TD6.
  • the audio data streams received from TD3 to TD5 make a non-zero contribution, but there is no contribution from the audio data stream received from TDl, TD2 or TD6, in the mixed audio data stream sent to TD6.
  • the audio data stream from TDl is sent, unmixed, to the other participant in the private conversation, TD2, and the audio data stream from TD2 is sent, unmixed, to TD2, during this period.
  • the second respective volume adjustment levels comprise relatively high, substantially equal, volume adjustment levels applied to each of the audio data streams received from TD3 to TD5 and the audio data streams received from TDl and TD 2 are cut out in the mixed audio data stream sent to TD6.
  • the mixed media stream can be seen to remain at a relatively low level, for example in the region of 0.4 to 0.9, in this embodiment 0.6.
  • the audio data streams from TD1 and TD2 may be re-introduced in the mixed audio data stream sent to TD6 in response to detecting an utterance of a word or phrase, for example "end private" during said private telephone call indicating that the user of TD1 wishes to end the private telephone call with the user of TD2.
  • TD1 may for example be a desktop telephone and computing device 114 may be a desktop personal computer.
  • ATS 100 can provide information about a multi-party audio teleconference call being conducted between TD1 and other telephony devices, for example TD2 to TD6.
  • ATS 100 provides a log-in web page via its web interface 210. The user of TD1 is able to log-in to the web page, for example by entering the telephone dialling number of TD1 or other appropriate identifier which ATS 100 will recognise as being associated with the multi-party audio teleconference call being conducted between TD1 and TD2 to TD6.
  • the web interface 210 allows display of information associated with the multi-party audio teleconference such as visual indicators for each audio teleconference call participant.
  • the web interface 210 allows a user to configure various settings such that the user can configure multi-party audio teleconference services provided via ATS 100.
  • Example of such configurable settings include the predetermined time period, the predetermined threshold volume adjustment level and the one or more predetermined words described above in relation to multi-party audio teleconference control embodiments.
  • the web interface 210 allows triggering for switching between mixing modes. For example, instead of a user having to raise their voice in order for the second audio mixing mode to be employed to generate a mixed audio data stream, the user may instead enter appropriate user input via the web interface to instruct ATS 100 manually. Further, a user may manually trigger a direct conversation with another user (with associated boosting of their associated audio data stream in the mixed audio data stream), for example by clicking on a visual indicator associated with that user in the web interface 210. Similarly, a user may manually trigger a private conversation with another user (with their respective audio data stream not being combined into mixed audio data streams transmitted to the remaining participants of the multi-party audio teleconference call), for example by clicking on a further visual indicator associated with that user in the web interface 210.
  • the multi-party teleconference call is between a total of six participants, it should be understood that any number of participants may be accommodated within a teleconference, and that the audio mixing modes shown may have any number of participants within a particular mixing mode.
  • a mixing mode in which one participant is emphasized, and a mixing mode in which one or more other participants are de- emphasized (but still heard) may be provided, similar to the embodiment shown in Figure 4.
  • a mixing mode in which two or more participants are emphasized, and a mixing mode in which one or more other participants are de-emphasized (but still heard) may be provided, similar to the embodiment shown in Figure 5.
  • a mixing mode in which two or more participants having a private chat are cut out, and in which one or more other participants are relatively emphasized (compared to a default mixing mode) may be provided, similar to the embodiment shown in Figure 6.
  • telephony device TD1 could comprise a mobile telephony device and telephone switch 104 could comprise a mobile switching centre in a mobile telecommunications network connected to network 102.
  • telephone switch 104 could comprise a mobile switching centre in a mobile telecommunications network connected to network 102.
  • display and user input capabilities of computing device 114 instead of the display and user input capabilities of computing device 114 being employed to interface with the web server interface of ACS 100, corresponding display and user input functionality of the mobile telephony device could be employed.
  • Embodiments could be implemented on a mobile telephony device as an application installed on the mobile telephony device.
  • telephony device TD1 could comprise a smart desk phone.
  • telephony device TD1 could comprise a smart desk phone.
  • display and user input capabilities of computing device 114 instead of the display and user input capabilities of computing device 114 being employed to interface with the web server interface of ACS 100, corresponding display and user input functionality of the smart desk phone could be employed.
  • embodiments are applied to audio teleconference services; however it should be appreciated that embodiments may also be applied in relation to other teleconference services, such as video teleconference services.

Landscapes

  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • General Engineering & Computer Science (AREA)
  • Telephonic Communication Services (AREA)

Abstract

Dans la présente invention, une première branche d'appel adaptée pour transmettre des données audio associées à une téléconférence téléphonique multi-partie est établie entre un premier dispositif téléphonique et un service de téléconférence multi-partie. Le service de téléconférence multi-partie selon l'invention est apte à connecter le premier dispositif téléphonique à au moins deux autres dispositifs téléphoniques durant la téléconférence téléphonique multi-partie. Des données d'état indiquant que la téléconférence téléphonique multi-partie est d'un second type d'appel sont enregistrées. Durant la téléconférence téléphonique multi-partie, une demande d'établissement de communication téléphonique entrante pour le premier dispositif téléphonique est reçue. Cette demande est associée à une communication téléphonique entrante d'un premier type d'appel. Elle est reçue d'un autre dispositif téléphonique qui ne prend pas part à la téléconférence téléphonique multi-partie. La demande d'établissement de communication téléphonique entrante et la téléconférence téléphonique multi-partie sont gérées conformément aux données d'état enregistrées qui indiquent que la téléconférence téléphonique multi-partie est du second type d'appel.
PCT/GB2012/051433 2011-06-21 2012-06-21 Procédés et systèmes de téléconférence multi-partie Ceased WO2012175964A2 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
GB1110494.0 2011-06-21
GB1110494.0A GB2492103B (en) 2011-06-21 2011-06-21 Multi party teleconference methods and systems

Publications (2)

Publication Number Publication Date
WO2012175964A2 true WO2012175964A2 (fr) 2012-12-27
WO2012175964A3 WO2012175964A3 (fr) 2013-03-21

Family

ID=44454411

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/GB2012/051433 Ceased WO2012175964A2 (fr) 2011-06-21 2012-06-21 Procédés et systèmes de téléconférence multi-partie

Country Status (2)

Country Link
GB (1) GB2492103B (fr)
WO (1) WO2012175964A2 (fr)

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN108076013A (zh) * 2016-11-14 2018-05-25 展讯通信（上海）有限公司 一种多通路终端多方通话方法及装置
CN108124243A (zh) * 2016-11-29 2018-06-05 展讯通信（上海）有限公司 一种多通路终端多方通话方法及装置
CN108206898A (zh) * 2016-12-20 2018-06-26 展讯通信（上海）有限公司 实现多方通话的方法、装置及多通终端
CN109246386A (zh) * 2018-11-22 2019-01-18 湖北安心智能科技有限公司 一种无纸化数字会议系统及控制方法
US12143428B2 (en) 2022-11-30 2024-11-12 T-Mobile Usa, Inc. Enabling a wideband codec audio call between a mobile device and a wireless telecommunication network support center

Families Citing this family (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE102022200416A1 (de) * 2022-01-14 2023-07-20 Heinlein Support GmbH Kontrollverfahren für eine Kontrolle einer virtuellen Podiumsdiskussion über eine Kommunikationsverbindung zwischen einer Vielzahl von Kommunikationsteilnehmern

Family Cites Families (16)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB2284968A (en) * 1993-12-18 1995-06-21 Ibm Audio conferencing system
JPH1075310A (ja) * 1996-08-29 1998-03-17 Nec Corp 多地点テレビ会議システム
CA2242426A1 (fr) * 1997-07-07 1999-01-07 Northern Telecom Limited Methode et appareil de teleconference
EP1038384B1 (fr) * 1997-12-12 2004-10-06 Thomson Consumer Electronics, Inc. Procede et systeme permettant de repondre a un appel entrant pendant une conference telephonique
US6501739B1 (en) * 2000-05-25 2002-12-31 Remoteability, Inc. Participant-controlled conference calling system
US6792092B1 (en) * 2000-12-20 2004-09-14 Cisco Technology, Inc. Method and system for independent participant control of audio during multiparty communication sessions
US20060023900A1 (en) * 2004-07-28 2006-02-02 Erhart George W Method and apparatus for priority based audio mixing
JP2006148816A (ja) * 2004-11-24 2006-06-08 Nec Corp 中継制御装置、多地点間会議システム及び多地点間会議方法
US7734692B1 (en) * 2005-07-22 2010-06-08 Oracle America, Inc. Network collaboration system with private voice chat
JP4483779B2 (ja) * 2005-12-28 2010-06-16 ブラザー工業株式会社 割り込み着信接続方法及びコードレス電話装置
US20070263603A1 (en) * 2006-04-17 2007-11-15 Lucent Technologies, Inc. VoIP PERSONAL CALL RECORDER
US8670537B2 (en) * 2006-07-31 2014-03-11 Cisco Technology, Inc. Adjusting audio volume in a conference call environment
US8654954B2 (en) * 2007-01-03 2014-02-18 Alcatel Lucent System and method for controlling access to conference calls
US20080165708A1 (en) * 2007-01-08 2008-07-10 Avaya Technology Llc Multimedia conferencing method and signal
US20090097625A1 (en) * 2007-10-15 2009-04-16 Peters Mark E Method of and System for Controlling Conference Calls
US20110044474A1 (en) * 2009-08-19 2011-02-24 Avaya Inc. System and Method for Adjusting an Audio Signal Volume Level Based on Whom is Speaking

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
None

Cited By (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN108076013A (zh) * 2016-11-14 2018-05-25 展讯通信（上海）有限公司 一种多通路终端多方通话方法及装置
CN108076013B (zh) * 2016-11-14 2021-04-13 展讯通信（上海）有限公司 一种多通路终端多方通话方法及装置
CN108124243A (zh) * 2016-11-29 2018-06-05 展讯通信（上海）有限公司 一种多通路终端多方通话方法及装置
CN108206898A (zh) * 2016-12-20 2018-06-26 展讯通信（上海）有限公司 实现多方通话的方法、装置及多通终端
CN109246386A (zh) * 2018-11-22 2019-01-18 湖北安心智能科技有限公司 一种无纸化数字会议系统及控制方法
CN109246386B (zh) * 2018-11-22 2020-12-01 湖北安心智能科技有限公司 一种无纸化数字会议系统及控制方法
US12143428B2 (en) 2022-11-30 2024-11-12 T-Mobile Usa, Inc. Enabling a wideband codec audio call between a mobile device and a wireless telecommunication network support center

Also Published As

Publication number Publication date
GB2492103A (en) 2012-12-26
GB201110494D0 (en) 2011-08-03
WO2012175964A3 (fr) 2013-03-21
GB2492103B (en) 2018-05-23

Similar Documents

Publication Publication Date Title
US8867721B2 (en) Automatic mute detection
US7180997B2 (en) Method and system for improving the intelligibility of a moderator during a multiparty communication session
US7058168B1 (en) Method and system for participant control of privacy during multiparty communication sessions
US7742587B2 (en) Telecommunications and conference calling device, system and method
US6870919B2 (en) Mute status reminder for a communication device
EP1616433B1 (fr) Indication automatique de volume sonore insuffisant pour des participants a une conference telephonique
US9215409B2 (en) Systems and related methods for controlling audio communications between a relay service and an audio endpoint
EP1841192A1 (fr) Présence et activation de préférences dans un système de téléphonie de messagerie vocale instantanée
US8363808B1 (en) Beeping in politely
US7657007B2 (en) Method and apparatus for instant voice messaging
US20040116130A1 (en) Wireless teleconferencing system
US11089541B2 (en) Managing communication sessions with respect to multiple transport media
WO2012175964A2 (fr) Procédés et systèmes de téléconférence multi-partie
GB2578121A (en) System and method for hands-free advanced control of real-time data stream interactions
US8184790B2 (en) Notification of dropped audio in a teleconference call
US7885396B2 (en) Multiple simultaneously active telephone calls
US20070036330A1 (en) Call logging and call logging notification at telecommunications service provider gateway
US7046784B2 (en) Polite call waiting notification
US20090097625A1 (en) Method of and System for Controlling Conference Calls
US7974399B2 (en) Enhanced whisper feature
US20060098798A1 (en) Method to selectively mute parties on a conference call
US8284926B2 (en) Enterprise-distributed noise management
JPH066470A (ja) 構内交換電話システム

Legal Events

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

Ref document number: 12737857

Country of ref document: EP

Kind code of ref document: A2

NENP Non-entry into the national phase

Ref country code: DE

122 Ep: pct application non-entry in european phase

Ref document number: 12737857

Country of ref document: EP

Kind code of ref document: A2