CN1738410A - Videophone and method for increasing speech quality using it - Google Patents

Abstract

The invention discloses a picture-phone for solving the problem that the present technique can not improve the speech quality without decreased video quality, comprising: a PSTN function module, a H.323/SIP function module, a speech switching module and a speech receiving/playing device. Wherein, said PSTN function module is connected to the PSTN network via the telephone line to complete the functions of common telephone and supply the dialing, calling and answering of PSTN telephone; said H.323/SIP function module is connected to the IP network via the network lines to complete the calling, answering, video-voice encoding/decoding, video/voice transmitting-receiving and the video displaying base on the H.323/SIP protocol; said speech switching module is connected to said speech receiving/playing device, the PSTN function module and the H.323/SIP function module to realize the switching between the PSTN function module and the speech receiving/playing device or between the H.323/SIP function module and the speech receiving/playing device. In addition, said invention also provides a method for using said picture-phone to improve the speech quality.

Description

A kind of video telephone and utilize it to improve the method for voice quality

Technical field

The present invention relates to multimedia technology field, specifically refer to be applied to a kind of video telephone in the multimedia communication and utilize it to improve the method for voice quality.

Background technology

PSTN (Public Switch Telephone Network): public switched telephone network) technology is a most general basic voice communication technology, use PCM (Pulse Code Modulation: pulse code modulation) coding, it is stable to have voice, the measured advantage of matter, this The Application of Technology has been strained the distance in the world widely, has promoted human interchange and social development.But along with computer technology, development of internet technology and universal, multimedia communication is just more and more paid attention to, and is progressively promoted and use, and believes in the coming years and just can step into family's epoch.

In the prior art, multimedia communication can be divided into according to the type of when communication bearer network: arrowband and broadband two classes wherein make the multimedia visual phone based on broadband H323 (the multi-media communication standard based on packet network that ITU-T organizes to set up)/SIP (Session Initiation Protocol:IETF organize to set up towards Internet meeting and call signaling protocol) agreement be subjected to user's attention and favor owing to broadband network is more suitable for multimedia communication.

But because H323/SIP is based on the agreement of broadband IP network, so communication quality is subjected to network quality affects big, when network stabilization poor, occur when congested, radio phone terminal based on H323/SIP just can not provide the multimedia communication that has good stability, can occur such as voice quality problems such as sound delay are excessive, sound is unclear and background sound is noisy, and then have a strong impact on the reception and registration of user speech information.

For addressing the above problem, prior art often adopts following two kinds of technical schemes:

First kind: reduce video bandwidth automatically, promptly in multimedia communication H.323/SIP, when network occurs when unstable or congested by reducing the bandwidth of video section automatically, make the bandwidth of audio-frequency unit obtain raising to a certain degree, and then improve the quality and the stability of voice in the multimedia communication.

In multimedia communication based on agreement H.323/SIP, audio code stream and video code flow are that (Real-time Transport Protocol: agreement RTP), (User DatagramProtocol: User Datagram Protoco (UDP)) Bao form sends with UDP according to RTP.Because the UDP bag is subjected to the influence of network quality big, during network congestion, the UDP bag is easy to lose, thereby it is poor to look audio quality, and when network was unobstructed, UDP lost less or not and loses, thereby looked the audio quality height.By detecting the packet loss of audio code stream and video code flow, can judge the Congestion Level SPCC of network.

It is big that packet loss becomes, and the possibility network congestion is described, terminal and opposite end just cooperatively interact and reduce video bandwidth (size of video code flow just) automatically, thereby can reduce network congestion degree, improve network condition, reduce the audio code stream packet loss, in to a certain degree assurance audio quality.

Though first kind of scheme be the degree of alleviating network congestion to a certain extent, improve the quality of voice, but still have following shortcoming:

1, when improving voice quality, but make video quality descend greatly because of reducing video bandwidth automatically, i.e. the raising of voice quality is a cost to reduce video quality.

2, network quality is poor, and reason has multiple, in most cases can not reduce the data traffic of whole network from the individual reduction of speed of terminal, can not improve network condition, also just can not reduce the audio frequency packet loss and improve audio quality.

Second kind: use QOS (Quality of Service: service quality) technical scheme, promptly be to utilize the Priority flag of ether frame head to transmit video/audio code stream, utilize and support the router of QOS that the bag of high priority is preferentially sent, guarantee the transmission of the video/audio code stream of high priority.

Specific to based on the radio phone terminal of H323/SIP agreement the time, promptly be the priority of filling in by at the frame head of video/audio, make the transmission that when network congestion, guarantees the video/audio bag to guarantee audio quality.

Though second kind of scheme can reduce video quality, because QOS needs all-router support on the transmission path, so there are the following problems:

1, because the present not every router of network is all supported QOS, so can not support the enforcement of second kind of scheme effectively;

If 2 change the router of not supporting QOS in the network and support QOS, then need existing network is adjusted on a large scale, and then make cost increase greatly.

Summary of the invention

The object of the invention is to provide a kind of video telephone and utilizes its method that improves voice quality, to solve prior art can't improve voice quality under the situation that does not reduce video quality problem.

For addressing the above problem, the invention provides following technical scheme:

A kind of video telephone comprises: the PSTN functional module, and H.323/SIP functional module, voice handover module and voice reception/playing device, wherein,

Described PSTN functional module links to each other with the PSTN network by telephone wire, finishes common telephony functions, supports the pstn telephone dialing, calls out and answers;

Described H.323/SIP functional module links to each other with IP network by netting twine, finishes calling based on agreement H.323/SIP, answers, video/audio encoding and decoding, looks audio frequency and send and receive and video shows;

Described voice handover module respectively with described voice reception/playing device, PSTN functional module and H.323/SIP functional module link to each other, can be implemented in PSTN functional module and switching voice reception/playing device between the functional module H.323/SIP by described voice handover module.

Described voice reception/playing device is a microphone.

Described PSTN functional module has microphone interface and pstn telephone line interface.

Described H.323/SIP functional module has microphone interface, and display screen is arranged, and network interface is arranged, and wherein, described microphone interface is used for voice output and sound input; Described display screen is used for display video image; Described network interface is connected with netting twine, is used for accessing to wide band network.

Described voice handover module has by hands-free audio amplifier, hands-free MIC and the combination of hands-free commutation circuit.

A kind of method of utilizing described video telephone to improve voice quality comprises the steps:

Monitoring step, in based on multimedia communication process H.323/SIP, any end all monitors voice quality;

Switch step, when any end monitors that the discovery voice quality is relatively poor, promptly under the situation that does not disconnect existing H.323/SIP conversation, utilize the PSTN functional module to call out the opposite end telephone number, simultaneously voice reception/playing device is switched to the PSTN functional module, make user speech be sent to the opposite end by PSTN network and PSTN functional module.

Described voice reception/playing device is a microphone.

Use technical scheme of the present invention, have following advantage:

1, by utilizing the PSTN network to carry out high-quality voice communication, and then solved effectively and called out H.323/SIP merely that voice quality is low down, be subjected to the big problem of web influence.

2, guaranteeing that voice communication quality is not subjected under the prerequisite of web influence, whether the user can also be according to network condition etc., freely select to use the PSTN network to carry out voice communication, as when network is more stable, just need not select the PSTN network to carry out voice communication; When unstable networks or obstruction, then select the PSTN network to carry out voice communication.

3, owing to can being chosen in the PSTN network, user in this programme carries out voice communication, so when the user does not need to carry out video communication when carrying out voice communication, just can only carry out PSTN and call out, and need not H.323/SIP call out, and then make the user that the bigger degree of freedom arranged.

Description of drawings

The video telephone module map of Fig. 1 in realizing the inventive method process, using among the embodiment;

Fig. 2 is the network connection diagram in realizing the inventive method process among the embodiment;

Fig. 3 after H.323/SIP calling out foundation earlier among the embodiment, sets up the schematic flow sheet that PSTN calls out again;

Fig. 4 is that PSTN calls out foundation earlier among the embodiment, sets up the schematic flow sheet of H.323/SIP calling out again;

Fig. 5 is for initiating the schematic flow sheet that H323/SIP, PSTN call out simultaneously among the embodiment.

Embodiment

Before the specific implementation process of the method for introducing raising voice quality of the present invention,, respectively video telephone modular structure and the network annexation in realizing the inventive method process are made a presentation earlier in conjunction with Fig. 1 and Fig. 2.

Please refer to Fig. 1, be the video telephone module map of using among the embodiment in realizing the inventive method process, described video telephone comprises: PSTN functional module, H.323/SIP functional module and voice handover module and microphone.

Described PSTN functional module links to each other with the PSTN network by telephone wire, finishes common telephony functions, supports the pstn telephone dialing, calls out and answers, and has microphone interface and pstn telephone line interface.The microphone interface of described microphone interface and plain old telephone is identical, is used for the input of voice output and sound; Telephone line interface is used for inserting ordinary telephone line to video telephone H.323/SIP.

Described H.323/SIP functional module links to each other with IP network by netting twine, finishes calling based on agreement H.323/SIP, answers, video/audio encoding and decoding, looks audio frequency and send and receive and video shows.Described H.323/SIP functional module is stated in realization in the process of function and can be realized according to agreement H.323, can realize that also perhaps two agreements all realize according to Session Initiation Protocol.Described H.323/SIP functional module has microphone interface, and display screen is arranged, and network interface is arranged, and wherein, described microphone interface is used for voice output and sound input; Described display screen is used for display video image; Described network interface is connected with netting twine, is used for accessing to wide band network.

H.323/SIP described voice handover module reaches with described microphone, PSTN functional module respectively, and functional module links to each other, can be implemented in the PSTN functional module and H.323/SIP switch microphone between the functional module by described voice handover module, when switching to the PSTN functional module, microphone links to each other with the microphone interface of PSTN functional module, microphone can receive and play the telephone sound signal that the PSTN module transmits like this, and (microphone: input transfer microphone) is to the PSTN functional module with microphone MIC.When switching to H.323/SIP functional module, microphone links to each other with the microphone interface of functional module H.323/SIP, microphone just can receive and play the H.323/SIP audio output signal of functional module like this, and the voice input signal of microphone MIC is sent to H.323/SIP functional module, and then through H.323/SIP transmitting to IP network behind the functional module coding.The voice handover module also can connect two modules simultaneously, like this voice of PSTN functional module and H.323/SIP the functional module voice can play simultaneously, the voice of PSTN functional module and H.323/SIP functional module also all send the voice that transmit by microphone MIC that send through the voice handover module.

Certainly the above-mentioned microphone MIC with voice reception/playing function can also substitute with the voice reception/playing device with said function in actual applications, because this class device has much in the prior art, just no longer specifically gives an example at this.

In addition, the voice handover module also can have the hand-free call function, and described hand-free call function is by hands-free audio amplifier, and hands-free MIC and hands-free commutation circuit make up and realize (not shown).

Please refer to Fig. 2, be the network connection diagram in realizing the inventive method process among the embodiment, in described network, video telephone A links to each other with PSTN network and IP network with video telephone B, that is: the voice communication between video telephone A and video telephone B can be finished by the PSTN network, can finish by IP network, also can finish by PSTN network and IP network simultaneously.Be that video telephone A and video telephone B can set up two callings simultaneously, PSTN calls out and H.323/SIP calls out, and wherein, H.323/SIP calls out and be responsible for transmitting video, data and network audio between two video telephone; PSTN calls out and then is responsible for transmitting the pcm encoder voice.

Core of the present invention is following two steps: monitoring step, that is: and in based on multimedia communication process H.323/SIP, any end all monitors voice quality; Switch step, promptly, when any end monitors that the discovery voice quality is relatively poor, promptly under the situation that does not disconnect existing H.323/SIP conversation, utilize the PSTN functional module to call out the opposite end telephone number, simultaneously voice reception/playing device is switched to the PSTN functional module, make user speech be sent to the opposite end by PSTN network and PSTN functional module.Because the quality of the voice that transmit by the PSTN network is far superior to the quality of the voice that transmit based on agreement H.323/SIP, and be not subjected to the influence of network congestion, so the present invention can solve the problem that descends based on voice quality in the agreement transport process H.323/SIP effectively; Simultaneously, because the bandwidth that the present invention does not also reduce also makes the bandwidth of video increase to some extent when forwarding the voice transmission transmission of to PSTN network, so the present invention has also improved the delivery quality of video to a certain extent on the contrary.Below in conjunction with Fig. 3, Fig. 4 and Fig. 5 method of the present invention is described in detail:

Before concrete the introduction, be defined as follows earlier:

H.323/SIP call out, being is the request of initiating to set up communication by agreement H.323 to the other side;

PSTN calls out, and is by the PSTN network and dials the other side to establish a communications link.

Please refer to Fig. 3, after H.323/SIP calling out foundation earlier among the embodiment, set up the schematic flow sheet that PSTN calls out again, described flow process is specially:

One, H.323/SIP setting up between the functional module of video telephone A and video telephone B called out, and sends video/audio code stream mutually, and image is presented at display screen, and the voice handover module of functional module plays back sound by switching to H.323/SIP;

Two, call out the pstn telephone number of video telephone B with the PSTN functional module of video telephone A; Simultaneously the voice handover module of video telephone B switches to PSTN functional module and functional module admixture H.323/SIP with microphone, if switch to voice certainly now immediately, then both sides can't exchange voice in the PSTN calling procedure, because calling out also, do not set up PSTN this moment, after the voice handover module switches to the PSTN functional module, can't hear the other side's voice, though promptly H323/SIP has set up and can lead to voice, the tube of not answering;

Three, called video telephone B answers pstn telephone, the voice handover module of video telephone B switches to the PSTN functional module, microphone is heard is speech from the opposite end video telephone A that PSTN transmits, video telephone A also can hear the voice of video telephone B from microphone, continue to keep and H.323/SIP call out;

Four, behind the PSTN call through, the video telephone A of caller switches to the PSTN functional module with the voice handover module; And then make video telephone A and video telephone B just can use the logical voice of PSTN functional module, H.323/SIP functional module intervisibility frequency and data.

Please refer to Fig. 4, set up for PSTN among the embodiment calls out earlier, set up the schematic flow sheet of H.323/SIP calling out again, the detailed process of described flow process is:

One, when video telephone A and video telephone B is in the PSTN communication process, on voice handover module separately all switches to separately PSTN functional module from separately H.323/SIP functional module;

Two, video telephone A initiates H.323/SIP to call out;

Three, video telephone B receipt of call sends video code flow, prepares decoding opposite end code stream and demonstration simultaneously;

Four, video telephone A finds H.323/SIP to exhale logical, sends video code flow, prepares decoding opposite end code stream simultaneously and shows, and then make that video telephone A and video telephone B can be with the logical voice of PSTN functional module, and H.323/SIP the functional module intervisibility frequently and data.

Please refer to Fig. 5, for initiating the schematic flow sheet that H323/SIP, PSTN call out among the embodiment simultaneously, described flow process is specially:

One, video telephone A initiates the PSTN calling and H.323/SIP calls out;

Two, arbitrary when arriving first video telephone B in PSTN calls out and H.323/SIP calls out, video telephone B just accepts the calling that arrives first; If two callings arrive simultaneously, then video telephone B freely selects a calling;

If three, video telephone B accepts the PSTN calling earlier, then the voice handover module switches to the PSTN functional module, after accepting H.323/SIP calling, shows the video that is sent by video telephone A at display screen more then; H.323/SIP call out if accept earlier, then the voice handover module switches to H.323/SIP functional module earlier, then after accepting the PSTN calling, the voice handover module switches to the PSTN functional module again, and then make video telephone A and video telephone B can use the logical voice of PSTN functional module, H.323/SIP functional module intervisibility frequency and data.

Use technical scheme of the present invention, have following You point:

1, by utilizing the PSTN network to carry out high-quality Yu Yin communication, and then You Xiao ground has solved merely H.323/SIP it is low to call out Xia the Yu sound quality, inscribed by the big Wen of network Ying Xiang.

2, Zai guarantee voice communication quality be not subjected to network Ying Xiang before put, the Yong family can also be according to network condition Deng, whether Yong PSTN network carries out Yu Yin communication to Zi You Xuan Ze, as when network is more stable, just not Yong Xuan Ze PSTN network carries out Yu Yin communication; When unstable networks or obstruction, Ze Xuan Ze PSTN network comes Carry out Yu Yin communication.

3, Yu Yin communication can be carried out by Xuan Ze Zai PSTN network in You Yu this programme Zhong Yong family, so when Yonging the family When Xu Zai does not carry out carrying out video communication when Yu Yin communicates by letter, just can carry out the PSTN calling by Zhi, and need not H.323/SIP call out, and then the bigger Zi You degree that made handy family You.

Claims (7)

1, a kind of video telephone is characterized in that comprising: the PSTN functional module, and H.323/SIP functional module, voice handover module and voice reception/playing device, wherein,

Described PSTN functional module links to each other with the PSTN network by telephone wire, finishes common telephony functions, supports the pstn telephone dialing, calls out and answers;

Described H.323/SIP functional module links to each other with IP network by netting twine, finishes calling based on agreement H.323/SIP, answers, video/audio encoding and decoding, looks audio frequency and send and receive and video shows;

Described voice handover module respectively with described voice reception/playing device, PSTN functional module and H.323/SIP functional module link to each other, can be implemented in PSTN functional module and switching voice reception/playing device between the functional module H.323/SIP by described voice handover module.

2, video telephone as claimed in claim 1 is characterized in that, described voice reception/playing device is a microphone.

3, video telephone as claimed in claim 2 is characterized in that, described PSTN functional module has microphone interface and pstn telephone line interface.

4, video telephone as claimed in claim 2 is characterized in that, described H.323/SIP functional module has microphone interface, display screen and network interface, and wherein, described microphone interface is used for voice output and sound input; Described display screen is used for display video image; Described network interface is connected with netting twine, is used for accessing to wide band network.

As arbitrary described video telephone in the claim 1 to 4, it is characterized in that 5, described voice handover module has by hands-free audio amplifier, hands-free MIC and the combination of hands-free commutation circuit.

6, a kind of method of utilizing the described video telephone of claim 1 to improve voice quality is characterized in that described method comprises the steps:

Monitoring step, in based on multimedia communication process H.323/SIP, any end all monitors voice quality;

Switch step, when any end monitors that the discovery voice quality is relatively poor, promptly under the situation that does not disconnect existing H.323/SIP conversation, utilize the PSTN functional module to call out the opposite end telephone number, simultaneously voice reception/playing device is switched to the PSTN functional module, make user speech be sent to the opposite end by PSTN network and PSTN functional module.

7, method as claimed in claim 6 is characterized in that, described voice reception/playing device is a microphone.

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