KR100861590B1 - Internet telephone improving sound quality of voice communication and audio service of communicationtype - Google Patents

Abstract

The present invention relates to an internet telephone for providing a voice call and a call-type audio service of the original sound level.

According to the present invention, in an Internet telephone in which a call connection is made by a signaling protocol and a call is made based on the Internet network, in the case of a voice signal, whether or not the broadband codec is included in the codec information of the session description protocol (SDP) when the call is connected. A dual coder unit configured to apply a narrowband codec or a wideband codec and apply a predetermined audio codec separately from the voice signal in the case of an audio signal; An audio mixing unit for mixing or separating the audio signal and the audio signal output through the dual coder unit; An audio switching unit for determining an output path of the audio signal and the audio signal output through the dual coder or the audio mixing unit, and a series for receiving voice calls and audio services through the Internet by controlling organic operations of the above-described components. It includes a control unit for performing the processing of.

Therefore, the sound quality of voice and audio data is improved to the original sound level when providing a call-type audio service such as a voice call which is an original function of a phone, a call connection sound, and a background sound during a call through the Internet network.

Internet Phone, Call Audio Service, Dual Coder

Description

INTERNET TELEPHONE IMPROVING SOUND QUALITY OF VOICE COMMUNICATION AND AUDIO SERVICE OF COMMUNICATIONTYPE}

1 is a block diagram showing an essential part of an internet telephone for providing a voice call and an audio service during a call according to an embodiment of the present invention.

2 is a block diagram showing the overall configuration of an internet telephone for providing a voice call and an audio service during a call according to an embodiment of the present invention.

3 is a diagram illustrating an audio switching unit of an internet telephone for providing an audio call and an audio service during an original sound level according to an embodiment of the present invention.

4 is a view showing a call flow as an embodiment according to the present invention.

The present invention relates to an internet telephone for providing a voice call and a call-type audio service of the original sound level, and more particularly, provides a call-based audio service such as a call connection sound, a background sound during a call, and a sound quality of the original sound level. It is about Internet telephone.

With the widespread use of the Internet, Voice over Internet Protocol (VoIP) technology for transmitting voice signals through the Internet has attracted much attention. Internet telephony technology, called Voice Over Internet Protocol (VoIP), is a series of communication service technologies that convert voice data into Internet protocol data packets on a computer network to enable voice calls, such as telephone calls in a general telephone network.

The telecom market is integrating voice, data and video via Internet protocols. Voice over Internet Protocol (VoIP) technology has recently been in the spotlight due to the reduction of communication charges through efficient use of existing network resources and the ease of management through integration of telephone and data networks. In addition, fixed-line operators are spurring the development of services for the Internet telephony market in accordance with the Ministry of Information and Communication's decision to grant incoming VoIP calls.

On the other hand, related services focused on various telephone supplementary services, which are non-speech sectors, have been developed in various ways in the wireline telephone industry, which was considered cumbersome and inconvenient due to the development of mobile communication and diversification of mobile phone functions. This is a reflection of the fact that value-added services are becoming more and more important given the slump in utilization of fixed-line phones due to the market saturation and the increase in mobile phone subscribers.

In particular, in the wireline industry, the competitiveness of value-added services rather than the classic voice service has been a factor in securing subscribers after full implementation of local phone number portability. As cable operators aiming at maximizing profits are strengthening the intelligent network segment, related service launches are becoming a hurdle, while fixed-line phones, which only deliver voice, are releasing various additional services using intelligent network services such as toll-free telephone and lifetime number.

However, in the conventional communication environment through the circuit telephone network, it is necessary to consider the limited bandwidth of the circuit telephone network and the transmission quality according to the line characteristics of the analog method, so that only the narrowband information that can be transmitted to the circuit telephone network is sampled. Delivered. Therefore, the entire signal could not be transmitted. Of course, even if only the narrowband information is sampled and delivered in the case of a voice signal, the sensitivity of the listener to the loss is not high. However, in the case of an audio signal based on an additional service, if only the narrowband information is sampled and delivered, the listener is not aware of the loss. The degree of sound quality you feel is considerable.

This is pointed out as a very big problem that causes the dissatisfaction of the subscribers due to the deterioration of the audio service quality due to the sound quality in the situation that various applications or application services using the telephone network are bursting.

Therefore, in order to reflect the current situation in the industry that provides various audio services in addition to voice calls, solving the problem of deterioration of sound quality for audio sources is a very big problem in communication services.

Therefore, the technical problem to be achieved by the present invention is to provide a technology for realizing the sound quality of the original sound level in providing a call-type audio service, including a voice call that is a unique function of the phone through the Internet phone that makes a call over the Internet network. .

Internet phone according to the characteristics of the present invention for achieving the above-described technical problem, the call connection is made by the signaling protocol and the call is based on the Internet network, in the case of a voice signal in the call connection session description protocol A dual coder unit for applying a narrowband codec or a wideband codec according to whether the wideband codec is included in the codec information of the SDP, and applying a predetermined audio codec separately from the voice signal in the case of an audio signal; An audio mixing unit for mixing or separating the audio signal and the audio signal output through the dual coder unit; An audio switching unit for determining an output signal of the audio signal and the audio signal output through the dual coder or the audio mixing unit and the voice call over the Internet by controlling the interaction of the dual coder unit, the audio mixing unit and the audio switching unit. And a control unit for performing a series of processes for receiving an audio service,
The control unit sets the voice channel and the audio channel to separate channels that do not interfere with each other, and sets the transmission and reception of voice and audio signals through the respective channels.

In this case, the dual coder unit includes at least one of G.723.1, G.729, G.711 M-law, and G.711 A-law as a narrowband codec, and includes G.722 as a wideband codec.

In addition, the dual coder unit includes an MP3 and a WMA codec as an audio codec.

In addition, the dual coder unit preferably includes an echo cancellation unit for canceling an echo with respect to a signal processed by a narrowband or wideband codec.

On the other hand, the Internet phone according to the present invention is an input / output means for providing an input / output interface of the audio signal and the audio signal handset for handset with the other party consisting of a microphone and a speaker for an audio device; A speakerphone function unit including a microphone for an audio device and stereo speakers composed of two speakers to divide and output a signal output through the dual coder unit or the audio mixing unit; And consisting of a port for connecting to the external speaker includes an external speaker connection for dividing the sound output.

In this case, the control unit includes an audio signal configured to start an audio channel or start an audio channel only while the voice channel is started.

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The controller may be configured to start an audio channel by setting 'User-Info' as a content ID in 'Request-URI' of the call connection message and 'S Internet-URI in the To header'.

DETAILED DESCRIPTION Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings so that those skilled in the art may easily implement the present invention. As those skilled in the art would realize, the described embodiments may be modified in various different ways, all without departing from the spirit or scope of the present invention. In the drawings, parts irrelevant to the description are omitted in order to clearly describe the present invention, and like reference numerals designate like parts throughout the specification.

In addition, when a part is said to "include" a certain component, it means that it may further include other components, without excluding other components unless otherwise stated.

In addition, the term module described herein refers to a unit for processing a specific function or operation, which may be implemented in hardware or software, or a combination of hardware and software.

Now, with reference to the drawings will be described in detail with respect to the Internet phone for providing a voice call and a call-type audio service of the original sound level according to an embodiment of the present invention.

First, when the internet phone according to the embodiment of the present invention is connected by the signaling protocol in the two phone periods and the media transmission is started, the received packet-type signal is restored by the codec unit and the D / A conversion, the handset Or, since the voice is output to an output device such as a speakerphone, an internet phone technology, commonly referred to as voice over internet protocol (VoIP) technology, in which voice calls are made, is adopted.

At this time, the connection establishment protocol uses Session Initiation Protocol (SIP) as a main protocol means, and processes incoming / outgoing signals by the session establishment protocol, and an audio service including an additional communication function uses an extended session establishment protocol. In this case, media transmission is performed by Real-time Transport Protocol (RTP). However, it can be configured to cope with other connection protocols of VoIP protocols such as H.323 and MGCP, and supports both Internet Protocol V4 and Internet Protocol V5 to prepare for future changes in Internet equipment. It is preferable.

According to an exemplary embodiment of the present invention, as shown in FIG. 1, the network interface unit 100 receives a voice signal and an audio signal through an internet network, and the dual coder unit 200 adapts the voice codec according to the characteristics of the received signal. And audio codec.

In this case, the dual coder 200 applies a codec corresponding to the narrow codec or the wide codec according to whether the received voice signal is a narrowband signal or a wideband signal.

The audio mixing unit 300 mixes the audio signal and the audio signal output through the dual coder unit.

The audio switching unit 400 selects a path for outputting a voice signal or an audio signal output through the dual coder 200 or the audio mixing unit 300, or a signal mixed with the voice signal and the audio signal. do.

In addition, the input / output unit 500 outputs the voice signal and the audio signal, the mixed voice and the audio signal from the dual coder 200 or the audio mixing unit 300 by the selection of the audio switching unit 400.

Referring to Figure 2 will be described in more detail the overall configuration of the Internet phone for providing a voice call and a call-type audio service of the original sound level according to an embodiment of the present invention.

The network interface unit 100 is a means for interfacing an internet phone with an internet network. The network interface unit 100 transmits / receives an incoming / outgoing signal of the Internet, converts a voice and audio signal converted into a packet into an electrical signal, and transmits or receives it to the Internet network. do. In addition, the audio and audio signals may be set to separate channels that do not interfere with each other, and the audio channel may be started or only the audio channel may be started alone while the voice channel is started. In this case, when only the audio channel is activated alone, the User-Info part is described as a content ID among the Request-URI of the invitation message and the S Internet-URI of the To header.

The dual coder unit 200 is responsible for changing, compressing, or releasing voice and audio signals transmitted or received through the network interface unit 100. In particular, the respective codecs are applied to the voice signal and the audio signal, and the corresponding codecs are applied to the voice signal by dividing the narrowband signal and the wideband signal. In this case, a narrowband codec is applied to a narrowband communication network voice signal such as a public switched telephone network (PSTN) or a mobile communication network. Broadband codec is applied to voice signal. The narrowband codec may include at least one of G.723.1, G.729, G.711 M-Law, and G.711 A-law, and the wideband codec may include G.722. In addition, the audio codec is applied to the audio signal, and it is preferable to include the MP3 and WMA codecs. Here, a detailed process of classifying the voice signal and the audio signal and applying the codec suitable for each characteristic by dividing the voice signal into the narrowband signal and the wideband signal will be described in detail with reference to FIG. 4. .

On the other hand, the audio mixing unit 300 is a means for providing a call background sound during a voice call, a means for mixing a voice signal and various audio sound sources. That is, when only the audio signal is received through the voice channel and the audio channel is started, the audio signal received through the voice channel and the audio signal received through the audio channel are mixed and provided to the user through the output means.

The audio switching unit 400 is a means for setting an output path of a voice and an audio signal output through the dual coder unit or a signal mixed with a voice and an audio signal output through the audio mixing unit. That is, it detects whether the handset is on / off, whether the external speaker is connected to the external speaker connection, etc., and selects the output path according to the user's selection including the background sound setting. As shown in FIG. 3, when outputting a voice signal separated from an audio signal, switch 1 (SW1) or switch 3 (SW3) is connected to a handset portion of a handset or a speaker portion of a speakerphone function portion. In addition, when outputting a mixture of audio and audio signals, switch 1 (SW1) or switch 3 (SW3) is connected to a digital analog converter (DAC), switch 2 (SW2) and switch 4 (SW4) Connect. That is, for example, during a call that does not include a call connection sound or an audio signal, the voice signal is outputted to the speaker of the handset or the speakerphone function unit by connecting the switch 1 (SW1) or the switch 3 (SW3) to the speaker. In the voice signal including the audio signal, that is, the mixed signal, switch 1 (SW1) or switch 3 (SW3) is connected to the DAC, and the signal passing through the DAC is connected to switch 2 (SW2) or switch 4 (SW4), respectively. The speaker is output to the handset of the handset or the speaker of the speakerphone function unit. In the case of a pure audio signal that does not include a voice signal, the pure audio signal is output to an external speaker connected to an external speaker connection through a DAC connected to switch 1 (SW1) or switch 3 (SW3). (Reference, DAC (Digital Analog Converter) in the drawings are collectively described as a control unit in the present invention)

The input / output means for providing the input / output interface of the audio signal and the audio signal provides the same function as the audio device even when the internet phone according to the present invention provides a call-type audio service in addition to the voice call. In view of performance, it has characteristics for audio equipment. That is, a handset in which a speaker that outputs a voice signal and a microphone that electrically converts a voice signal to make a call with the other party has characteristics for an audio device for minimizing sound quality loss when providing a broadband call and audio service. It consists of a microphone and a speaker. In addition, as a means for enabling a call without a handset, the speakerphone function unit is also composed of a microphone and a speaker having characteristics for an audio device, and the speaker is composed of two speakers to divide a sound output in a stereo type. In addition, the external speaker connection portion provided with a port for connecting with an external speaker is provided with a stereo function to separate the sound output to the left and right speakers connected.

The key input unit is composed of a plurality of key buttons. The key input unit is a means for inputting a phone number or a letter of a counterpart and selecting a function such as setting a background sound during a call, outputting an audio signal, and selecting an output path.

The display unit may output a graphic signal such as an image or an image as a means for providing an interface screen to the user, and it is preferable to use a liquid crystal display (LCD).

The control unit processes data in order to perform the entire series of operations of the Internet telephone according to the present invention, and controls the organic operation of the above-described components, and connects to the Internet network through a network interface unit for call processing for incoming / outgoing calls. In addition, it performs a process for receiving an audio service. In this case, the controller is set to a separate channel in which the voice channel and the audio channel do not interfere with each other, and is configured to transmit and receive voice and audio signals through respective channels. In the case of the audio signal, the audio channel is started while the voice channel is started or only the audio channel can be started. The 'User-Info' in the 'Request-URI' of the call connection message and 'S Internet-URI' of the To header 'Is set as the content ID, so the audio channel is started. In addition, the VoIP application that supports Internet phone calls stored in the storage unit, the SIP stack, the Management API that runs the network stack and application programs, and the Web that implements functions such as selecting an audio source for audio service by accessing a web server. It runs the related software for running internet phone and audio service such as server, channel managing channel information and audio codec related application. Here, the storage unit also stores the audio sound source, the user profile and the like received through the Internet network.

Now, the operation of the present invention including the above-described components will be described. 4 shows a call flow as one embodiment according to the present invention, which will be referred to for explaining the operation of the present invention.

Basically, the audio channel and the audio channel are separated and operate without interfering with each other's channels, and then the audio mixing unit mixes the audio signal and the audio signal as needed (i.e., when the user selects the background sound). do.

First, the ring back tone is established by establishing an initial session, and the PRACK (early-answer) (S200) for the 18X Session Progress (early-offer) (S100) and 200 OK (S300) for the ring back tone This is an audio channel part (A) for an additional part when compared with a call flow based on the existing session establishment protocol. In case of requesting a call for a ring back tone, the coder follows the protocol of Session Initiation Protocol (SIP), and audio services such as call background sound, MOD (Music On Demand), Internet-Radio, etc. It uses Call Setup Flow of SIP, Session Initiation Protocol. Here, Session Initiation Protocol (SIP) is a protocol for controlling a terminal, i.e., an internet phone, and identifies each other in a terminal, i.e., an internet phone, to communicate on the Internet, locates each other, and creates a multimedia communication session between them. Application level signaling protocol that specifies the procedure to delete, delete, or change.

In addition, the channel for the voice call and the channel for the audio service are separated from each other and have a call setup process independently. The sound source for each channel is an output path by the audio mixing unit and the audio switching unit described in FIG. Is determined and output.

At this time, the call connection sound is driven by the audio coder unit separately from the voice coder unit (narrowband coder unit or wideband coder unit) for voice call.

As above (i.e., the PRACK (early-answer) (S200) for the 18X Session Progress (early-offer) (S100) and (200 OK) (S300) for it) the ring back tone When the audio channel is set for (A) after receiving 200 OK (S400) from the called party and sending an ACK (S500) indicating that the voice channel has been formed, when the channel is ready to be used as a voice channel, the incoming real-time After checking the payload of the header in the Real-time Transport Protocol (RTP), check the payload of the coder received at 200 OK (S400), and then switch from the audio channel to the voice channel. A voice call is performed through the voice channel (B).

Now, a detailed process of selecting a narrowband codec and a wideband codec in a voice call will be described.

If the codec included in the codec information of the session description protocol (SDP) of the called party response message is the broadband codec, the broadband coder unit is used. The call is initiated by initiating a voice channel. Here, the Session Description Protocol (SDP) is a protocol for transferring media connection information between a telephone and a media server in a terminal, that is, a telephone period.

In addition, when receiving a call from the called party's point of view, if the wideband codec is found in the voice codec list described in the codec information of the session description protocol (SDP) of the calling party's response message, it is determined that the wideband coder is supported to determine the priority of the coder. The call is made by initiating a voice channel through the broadband coder unit, ignoring the protocol of the session description protocol (SDP).

That is, in the voice coder, whether the call is made to the narrowband coder or the call to the broadband coder depends on whether the wideband codec is included in the codec information list of the session description protocol (SDP).

In the case of a voice call, the coder list of the session description protocol (SDP) of the call connection message describes all narrowband, wideband, and audio codecs. If the codec included in the codec information of the session description protocol (SDP) in the 18X Session Progress (early-offer) (S801) or the ACK (200 OK) (S803) for the call connection message is a narrowband codec or a wideband codec, Recognizes a voice call with no ringing tone.

If the selected codec is an MP3 codec or a WMA codec, it is recognized as a voice call with a ring back tone. Therefore, the ring back tone is basically activated only when the ring back tone is set on the called party.

When the call request for the audio service is requested, the codec information of the session description protocol (SDP) of the call connection message is written only for the audio codec and not for the voice codec. In other words, only MP3 codec or WMA codec is sent.

In this case, when requesting only an audio service without a voice call, the user makes a request using a key input unit and a display unit, and describes the 'User-info' part of the 'To header' of the call connection message as a content identifier instead of the receiving telephone number. To send. In this case, since only the audio signal is received using the connected external speaker, it can perform a function as an audio device.

Accordingly, according to an embodiment of the present invention, the audio service can be provided even when not in a call, including receiving an audio service during or during a call connection such as a call connection sound or a background sound. It is possible to provide a new concept of telephone which is applied.

According to the above-described configuration, the present invention provides an internet audio telephone in which the functions of a telephone and an audio terminal are combined to provide an audio service such as a call connection sound, a background sound, etc. in addition to a voice call which is an original function of the telephone.

In particular, since the received signal is divided into a voice signal and an audio signal, a narrowband voice signal and a wideband voice signal, and an appropriate codec is applied thereto, the biggest problem is provided in providing an audio service in a conventional circuit telephone network by preventing the loss of sound quality. It is largely to solve the problem of poor sound quality pointed out.

Therefore, based on the packet network (or IP network) replacing the conventional circuit telephone network, various audio-based application services related to intelligent and high-quality sound and new business models are continuously emerging, and new audio of various additional functions and application functions It is expected that services can be provided at the original level.

Claims (8)

delete delete delete delete In the Internet phone where the call connection is made by the signaling protocol and the call is made based on the Internet network, In case of a voice signal, a narrowband codec or a wideband codec is applied to the codec information of the session description protocol (SDP) when the call is connected, and in the case of an audio signal, a predetermined audio codec is applied separately from the voice signal. Dual coder unit; An audio mixing unit for mixing or separating the audio signal and the audio signal output through the dual coder unit; An audio switching unit for determining an output path of an audio signal and an audio signal output through the dual coder or the audio mixing unit; And a controller for controlling a mutual operation of the dual coder, the audio mixing unit, and the audio switching unit to perform a series of processes for receiving a voice call and an audio service through the Internet. The control unit, An Internet telephone, characterized in that the voice channel and the audio channel are set as separate channels that do not interfere with each other to control transmission and reception of voice and audio signals through each channel. The method of claim 5, wherein the control unit In case of an audio signal, the audio channel is started while the voice channel is started, or only the audio channel is started. Internet phone including a. The method of claim 5, wherein the control unit It is set to start audio channel by setting 'User-Info' as content ID in 'Request-URI' and 'Internet header' of 'To header'. Internet phone including a. 6. The apparatus of claim 5, further comprising: a handset for handset with a counterpart comprised of a microphone and a speaker for an audio device as an input / output means for providing an input / output interface of the voice signal and the audio signal; A speakerphone function unit including a microphone for an audio device and stereo speakers composed of two speakers to divide and output a signal output through the dual coder unit or the audio mixing unit; And Internet phone including an external speaker connection portion consisting of a port for connecting to the external speaker to divide the sound output.

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