CN100440911C - A method for adjusting IP telephone volume - Google Patents

A method for adjusting IP telephone volume Download PDF

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Publication number
CN100440911C
CN100440911C CNB031039871A CN03103987A CN100440911C CN 100440911 C CN100440911 C CN 100440911C CN B031039871 A CNB031039871 A CN B031039871A CN 03103987 A CN03103987 A CN 03103987A CN 100440911 C CN100440911 C CN 100440911C
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voice
user
volume
module
dsp module
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Expired - Fee Related
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CNB031039871A
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Chinese (zh)
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CN1523864A (en
Inventor
毛沈军
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Huawei Technologies Co Ltd
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Huawei Technologies Co Ltd
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Abstract

The present invention relates to a method for adjusting the sound volume of an IP telephone, which comprises the following steps: a. a user sends a DTMF number comprising a command for adjusting the sound volume of the IP telephone to a DSP module in a voice single plate; b. the DSP module identifies the DTMF number and also informs a VOIP conversation module on the side of a main plate to read the DTMF number; c. after the VOIP conversation module reads the DTMF number, the command for adjusting the sound volume of the IP telephone is sent to the DSP module; d. the DSP module modifies a parameter for decoding coded voice data, the voice data is restored into a voice signal, and the voice signal is played for the user. The present invention causes the user to flexibly adjust the sound volume of the listened sound of the end of the user by operating the keys of a double-voice multi-frequency telephone set according to personal needs and personal habits by introducing a mechanism for adjusting the sound volume on the IC card telephone set on a voice router. The present invention achieves the personal participation of the user, and an individualization arrangement can become flexible and convenient.

Description

A kind of method of regulating the IP phone volume
Technical field
The present invention relates to communication technical field, specifically refer to a kind of method of regulating the IP phone volume.
Background technology
Because IP phone can be finished the transmission of voice messaging by means of communication network, it is increasing the type of service while that data communication network provides, also effectively improved the utilization rate of data communication network, can be for the user provide the ratio of performance to price higher voice communication service, so obtaining user's approval in recent years and using widely.
But, because the IP phone gateway equipment of different vendor, employed gain parameter difference when the speech data after quantizing is handled, and then make that same voice messaging is handled the volume of the resulting voice messaging in back by different IP phone gateway may be too high or too low, can not satisfy user's requirement well.
For addressing the aforementioned drawbacks, prior art solutions is: each port of voice veneer on voice router, two orders of regulating volume are provided, and one is the adjusting command of input volume, zooms in or out with the speech volume to the input port input and controls; Another is the output volume adjusting order, with the control that zooms in or out of the volume of voice that receiving terminal is received.As seen, the volume of using above-mentioned two orders can regulate the IP phone voice that receive and send.
Though the size of above-mentioned technology scalable volume, but carrying out described two orders the time must use to be connected to the control terminal on the voice gateways or to be remotely logged on the voice gateways and manually be configured, and in case in the configuration, the voice gain parameter of this port is just fixing, that is: all be to use identical parameter to handle with any voice gateways conversation, the user can not be according to individual demand flexible volume.As seen, how designing a kind of user can own system or the method for controlling volume be the problem that the present utmost point of industry need solve.
Summary of the invention
The invention provides a kind of method of regulating the IP phone volume, can not be with what exist in the solution prior art according to the problem of user's actual needs flexible speech volume.
For achieving the above object, the present invention proposes a kind of method of regulating the IP phone volume, this method comprises following steps:
A, user send the DTMF that contains the instruction of adjusting IP phone volume, and (Dual ToneMulti-Frequency: dual-tone multifrequency) number is given DSP (the Digital Signal Processor: module digital signal processor) in the voice router voice veneer;
B, DSP module are discerned this DTMF number and are notified the VOIP of voice router mainboard side (speech that transmits on the Voice Over IP:IP) conversation module to read this DTMF number;
After c, this VOIP conversation module read this DTMF number, issue and regulate the IP phone volume and instruct this DSP module;
D, this DSP module are revised and are used for parameter that the speech data after being encoded is decoded, this speech data is reduced to voice signal plays to the user.
Wherein, regulating the instruction of IP phone volume among the described step a is meant: by the adjusting IP phone volume that the is combined to form instruction of several buttons on the double tone multiple frequency telephone set.
To read this DTMF number be to finish by the public mailbox that this VOIP conversation module and this DSP module can be visited to this VOIP conversation module among the described step c.
The present invention is by having introduced the mechanism of IC Card Telephone machine adjusted volume on voice router, make the size of the speech volume that the user can listen to according to the button flexible local terminal of individual human needs and custom operation double tone multiple frequency telephone set, realized the participation of individual subscriber, made personalized setting more flexibly, conveniently.
Description of drawings
Fig. 1 is the system architecture diagram of the IP phone data communication network of the embodiment of the invention;
Fig. 2 is the method flow schematic diagram of the embodiment of the invention.
Embodiment
The method of the described adjusting IP phone of embodiment of the invention volume is finished by voice router, the applied data communication network system figure of this method as shown in Figure 1, this network comprises: user A, user B and PBX (Private Branch eXchange: subscriber exchange) switch 11, first voice router 12, second voice router 13, GK server 14, IP network.
The interface single plate of the 2VI/4VI voice veneer that includes support voice encoding and decoding processing is installed on first voice router 12, second voice router 13 as required, and wherein: 2VI is meant the analog voice veneer of two-port; 4VI is meant the analog voice veneer of four ports.
The support of every voice veneer is respectively 2 or 4 ports, and the port type of support is FXS, FXO and E﹠amp; Three types of M, introduce the function of these three kinds of port types below respectively:
FXS interface finger print is intended user interface, be used for connecing the trunk interface of common simulation substation or switch, effect mainly provides line circuit interface, finish functions such as corresponding feed, ring, off-hook, on-hook, 2/4 line conversion, realization is to the access and the processing of analog voice signal, and the signal that will compress at last after handling is given main frame.
FXO interface finger print near-ring road trunk interface; be used for connecing the analog line of switch; effect mainly is corresponding ring detection, off-hook, protection, the conversion of 2/4 line an or the like function, realizes access and processing to analog voice signal, and the signal that will compress at last after handling is given main frame.
E﹠amp; M interface finger print is intended E﹠amp; The M trunk interface is used for connecing the E﹠amp of switch; The M trunk interface, effect mainly is the E﹠amp that finishes number of different types; The M signaling method is realized access and processing to analog voice signal, and the signal that will compress at last after handling is given main frame.
A DSP module is arranged respectively on each voice veneer, whether be used to discern actions such as user's off-hook, on-hook, dialing takes place, and can be by the VOIP conversation module of giving mainboard side with these event notices with mailbox public between the mainboard, simultaneously this DSP module can provide G.729A, G.723.1, the coding/decoding capability of type G.711.The embodiment of the invention is to be based upon on the basis of aforementioned prior art, and the DSP module that makes full use of on the voice veneer can be discerned the DTMF number, realizes on this VOIP conversation module.
Before the method for describing the described adjusting IP phone of embodiment of the invention speech volume size, set up the end-to-end voice channel of user A and B earlier, these are prior art, and it specifically comprises:
(1), telephone subscriber A plucked phone, the speech interface plate detects the off-hook action of user A, this signal is passed to the signal processing of mainboard side VOIP conversation module on second voice router 13.
(2), the VOIP conversation module is notified the DSP module playing alert tones on the voice veneer, the wait subscriber dialing.
(3), after the user dials phone number, these telephone numbers are collected and are stored by the VOIP conversation module.
(4), after collecting the target pattern that enough telephone numbers can mate a setting, telephone number or be mapped to an IP main frame by a dial plan mapper, perhaps on GK server 14, inquire about the pairing purpose IP of this telephone number address, i.e. first voice router 12 by second voice router 13.
(5), the VOIP conversation module is moved session protocol H.323 and set up a passage that sends and receive speech data for the connection that each has purpose IP address on IP network.
(6), the VOIP conversation module activates the encoding and decoding that connect two parties A, B, and uses RTP/UDP/IP as communication protocol stack.
Promptly finish the common talking state of user A, B by above step, hypothesis user A need regulate the IP phone volume below, is described the embodiment of the invention from amplifying the volume that local terminal answers and dwindling two aspects of volume that local terminal answers respectively below:
Embodiment one: amplify the volume that local terminal is answered
The first, the user sends and contains the DTMF number of regulating the instruction of IP phone volume and give DSP module in the voice veneer;
After user A uses IP phone and user B to enter communication process, when needs amplify the speech volume that local terminal answers, user's needs are double fast, and (the requirement telephone set must be the dual-tone multifrequency phone by " * " key on the telephone set, do not provide support for pulse telephone), the time of twice continuous button was controlled within two seconds.Why will be controlled at the time within 2 seconds is because " * " key may be employed the function button use of program as other, when the user uses IP phone, connectedly may require user A to carry out the process of interactive operation according to the information of voice prompt operation push-button to end subscriber B, the user may need at first to import " * " key or " # " key, " * " key or " # " key of this moment need send the opposite end to, or local terminal carries out some other processing, not as amplifying the function button that local terminal receives speech volume, the function button sequence of therefore using two quick continuous " * " keys to operate as amplification of volume.
The second, the DSP module is discerned this DTMF number and is notified the VOIP conversation module of mainboard side to read this DTMF number.
After the DSP module is discerned this DTMF number, the DTMF number that identifies is given in the public mailbox that DSP module and VOIP conversation module can visit, send an interruption to mainboard side simultaneously, the VOIP conversation module of notice mainboard side reads this DTMF number.
Three, after this VOIP conversation module reads this DTMF number, issue and regulate the IP phone volume and instruct this DSP module.
After the VOIP conversation module reads first " * " number button, at first apply for the timer in one two second, receive " * " number button merely and can not judge that the amplification of volume function key still needs the function key of other processing, therefore use timeout mechanism to distinguish.If within two seconds, receive the function key that " * " number button is just thought amplification of volume again, delete the timer in 2 seconds; If what receive is " # " number button, just think that rigidly connecting " * " number button of receiving is misoperation, neglect " * " number button, delete the timer in 2 seconds, then enter for the first time the processing procedure that receives " # " number button; If what receive is digital keys, just delete the timer in 2 seconds, directly according to " ' * ' number key+numerical key " carry out other processing; If in two seconds, do not receive any button, just think that this " * " number button is to need processed conventionally button, not the function button of regulating volume.
If in the time in two seconds, receive two " * " number buttons continuously, enter voice channel and regulate in the volume process, abandon this two buttons, after this whenever receive " * " number button, earlier " * " number key is discarded, notify voice veneer DSP module that sound is amplified simultaneously.
Four, this DSP module is revised and is used for parameter that the speech data after being encoded is decoded, again this speech data is reduced to voice signal and plays to the user.
The DSP module is just revised the parameter that decode operation uses, and it is regulated step-length and can be provided with, as regulating step-length 0.5dB or 1dB.Speech data behind the coding that the DSP module will be received from the IP network side joint uses that this amended parameter is decoded, play to user A after the D/A conversion, and local terminal has just received the voice that are exaggerated.In this process,, just withdraw from the process that the passage volume is regulated, with the digital normal process that receives if receive digital keys; If receive " # " number button, just handle according to receiving " # " number button for the first time, apply for the timer in one two second, wait for the common input that cooperates to come interpreting user of state of user key-press operation and timer.
Embodiment two: reduce the volume that local terminal is answered
After user A used IP phone to enter communication process with user B, when needs reduced speech volume that local terminal answers, the user needed double " # " key of pressing on the telephone set fast, and the time of twice continuous button also was controlled within two seconds.Its basic principle and embodiment one are similar, do not repeat them here.

Claims (3)

1, a kind of method of regulating the IP phone volume is characterized in that, this method comprises following steps:
A, user send to be contained the DTMF number of regulating the instruction of IP phone volume and gives DSP module in the voice router voice veneer;
B, DSP module are discerned this DTMF number and are notified the VOIP conversation module of voice router mainboard side to read this DTMF number;
After c, this VOIP conversation module read this DTMF number, issue and regulate the IP phone volume and instruct this DSP module;
D, this DSP module are revised and are used for parameter that the speech data after being encoded is decoded, this speech data is reduced to voice signal plays to the user.
2, a kind of method of regulating the IP phone volume as claimed in claim 1, it is characterized in that, regulate the instruction of IP phone volume among the described step a and be meant: by the adjusting IP phone volume that the is combined to form instruction of several buttons on the double tone multiple frequency telephone set.
3, a kind of method of regulating the IP phone volume as claimed in claim 1 is characterized in that, to read this DTMF number be to finish by the public mailbox that this VOIP conversation module and this DSP module can be visited to this VOIP conversation module among the described step c.
CNB031039871A 2003-02-18 2003-02-18 A method for adjusting IP telephone volume Expired - Fee Related CN100440911C (en)

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CN100440911C true CN100440911C (en) 2008-12-03

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Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR100650058B1 (en) * 2004-11-23 2006-11-27 주식회사 팬택 Method for controlling voice gain in a communication terminal and apparatus thereof
CN101207650B (en) * 2006-12-18 2010-05-19 中兴通讯股份有限公司 Method of volume control and silencing
CN101068289B (en) * 2007-06-21 2010-08-04 中兴通讯股份有限公司 Method and device for realizing automatic regulating speech sound volume in access-in gateway

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2000069155A2 (en) * 1999-05-05 2000-11-16 Nortel Networks Limited Telephony and data network services at a telephone
CN1282193A (en) * 1999-07-27 2001-01-31 三星电子株式会社 Method for adjusting honeycomb telephone communication voice and keycontrol tone volume
WO2001028256A1 (en) * 1999-10-14 2001-04-19 Conexant Systems, Inc. Method and apparatus for early detection of dtmf signals in voice transmissions over an ip network
KR20010090512A (en) * 2000-03-24 2001-10-18 김성철 Internet phone exchanger with gateway feature

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2000069155A2 (en) * 1999-05-05 2000-11-16 Nortel Networks Limited Telephony and data network services at a telephone
CN1282193A (en) * 1999-07-27 2001-01-31 三星电子株式会社 Method for adjusting honeycomb telephone communication voice and keycontrol tone volume
WO2001028256A1 (en) * 1999-10-14 2001-04-19 Conexant Systems, Inc. Method and apparatus for early detection of dtmf signals in voice transmissions over an ip network
KR20010090512A (en) * 2000-03-24 2001-10-18 김성철 Internet phone exchanger with gateway feature

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