CN113923198A - Method and device for improving low VoIP voice call quality - Google Patents
Method and device for improving low VoIP voice call quality Download PDFInfo
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- CN113923198A CN113923198A CN202111091555.8A CN202111091555A CN113923198A CN 113923198 A CN113923198 A CN 113923198A CN 202111091555 A CN202111091555 A CN 202111091555A CN 113923198 A CN113923198 A CN 113923198A
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- 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
- H04L65/80—Responding to QoS
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L15/00—Speech recognition
- G10L15/26—Speech to text systems
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L63/00—Network architectures or network communication protocols for network security
- H04L63/04—Network architectures or network communication protocols for network security for providing a confidential data exchange among entities communicating through data packet networks
- H04L63/0428—Network architectures or network communication protocols for network security for providing a confidential data exchange among entities communicating through data packet networks wherein the data content is protected, e.g. by encrypting or encapsulating the payload
Abstract
The invention provides a method and a device for improving low VoIP voice call quality. The method comprises the steps that a UAC establishes an SIP voice session through a UAS; the UAC detects the voice transmitted to a local microphone by a local user; the UAC performs character conversion on the detected voice through an STT analysis technology; the UAC sends the text to the UAS/remote user through SIP MESSAGE message; the remote user parses the received SIP MESSAGE message and displays the text message on a local display screen. In this way, the far-end receiver can see the SIP text message synchronously displayed on the local display screen while receiving and playing the voice, only a small amount of extra bandwidth is used, the understanding of the receiver to the voice is enhanced, the call quality is improved, the text is encrypted, and the information safety and the personal privacy protection are ensured.
Description
Technical Field
Embodiments of the invention relate generally to the field of communication networks, and more particularly to voice communication between SIP-based VoIP terminal devices.
Background
The IP-based voice transmission is a voice communication technology, and the basic principle is that voice data codes are compressed through a voice compression algorithm, then the voice data are packaged according to a TCP/IP standard, data packets are sent to a receiving place through an IP network, the voice data packets are concatenated, and the original voice signals are restored after decompression processing, so that the purpose of transmitting voice through the Internet is achieved. The core and key equipment of the IP telephone is an IP gateway which maps the area telephone area numbers of all areas into corresponding area gateway IP addresses. The information is stored in a database, and the data connection processing software can complete the functions of call processing, digital voice packaging, routing management and the like. When the user dials long distance call, the gateway determines the IP address of the corresponding gateway according to the telephone area code database data, and adds the IP address into the IP data packet, and selects the best route to reduce transmission delay, and the IP data packet reaches the gateway of the destination through Internet. In some areas where the Internet has not been extended or temporarily has no gateways installed, routes may be set up for switching over by the nearest gateway over the long distance telephone network to effectuate communication services.
Once the network environment has problems of network jitter, network speed reduction, packet loss and the like due to some reasons, the voice transmission based on the RTP/UDP must be affected. Although some voice data redundancy mechanisms (such as RFC2198/RFC6354) have been designed by RFC from the RTP level, which can compensate some voice quality loss to some extent in case of unstable network, the problem that the voice receiver cannot completely understand the semantics of the other party due to poor network quality still cannot be solved fundamentally.
Such as the patent: "VOIP conversation method, apparatus, system, device, and readable storage medium (CN 202010858125.3)": the application introduces a CDN acceleration network on the basis of the existing VOIP technology, namely, the CDN acceleration network consisting of a CDN control node and a plurality of CND acceleration nodes is used for helping the transmission of voice data between two call ends (an initiating end and a response end) and a media server. After the CDN acceleration network is introduced, the two call ends respectively transmit voice data to the media server indirectly through the CDN nodes corresponding to the two call ends, and by means of the characteristics of the CDN acceleration network, the media server has higher transmission speed, fewer intermediate routing times and stronger network fluctuation resistance, and brings better voice call quality. The scheme can only provide better call quality to a certain extent, and cannot fundamentally solve the problem of low call quality caused by network problems.
Disclosure of Invention
According to the embodiment of the invention, a method and a device for improving the low quality of VoIP voice call are provided.
In a first aspect of the invention, a method for improving VoIP voice call quality is provided. The method comprises the following steps:
s01: the UAC establishes an SIP voice session through the UAS;
s02: the UAC detects the voice transmitted to a local microphone by a local user;
s03: the UAC performs character conversion on the detected voice through an STT analysis technology;
s04: the UAC sends the text to the UAS/remote user through SIP MESSAGE message;
s05: the remote user parses the received SIP MESSAGE message and displays the text message on a local display screen.
Further, the SIP MESSAGE message sending and receiving conform to the RFC3428 standard and are performed simultaneously with the existing RTP message interaction.
Further, the UAC has a physical display screen.
Further, the UAC is supported by NPU.
Furthermore, the UAC display text interface is freely opened or closed by a user.
Furthermore, the physical display screen can display not less than 3 lines of characters, and each line displays not less than 10 characters.
Further, the S04 transmits SIP MESSAGE message using SIP-TLS.
Further, the S04 uses S/MIME mode to encrypt the text part separately.
Further, the S04 adopts a timing transmission mechanism, and the sender transmits the text converted from the speech at regular intervals.
Further, the said S04 adopts an intelligent sending mechanism, and the sender determines the sending timing and sending length by the escape sentence-breaking function of STT.
Further, the S05 adopts a timing display mechanism, and the receiving party displays the received text at regular intervals.
Further, the S05 displays the received text on a local display screen in real time.
Further, the S05 is to segment a SIP MESSAGE received text in a period and scroll the text on the display screen.
Furthermore, the text information is converted into a text and then stored locally, and a user manually turns pages to view the text information.
Further, the text is deleted in time after the call is completed.
In a second aspect of the present invention, an apparatus for improving VoIP voice call quality is provided. The device includes:
a connecting module: the UAC is used for establishing SIP voice session through the UAS;
a detection module: the UAC is used for detecting the voice transmitted to a local microphone by a local user;
a voice conversion module: the UAC carries out character conversion on the detected voice through an STT analysis technology;
a sending module: for UAC to send text to UAS/remote user via SIP MESSAGE message;
a receiving module: the remote user can analyze the SIP MESSAGE message and display the text information on the local display screen.
Further, the SIP MESSAGE message sending and receiving conform to the RFC3428 standard and are performed simultaneously with the existing RTP message interaction.
Further, the UAC has a physical display screen.
Further, the UAC is supported by NPU.
Furthermore, the UAC display text interface is freely opened or closed by a user.
Furthermore, the physical display screen can display not less than 3 lines of characters, and each line displays not less than 10 characters.
Further, the sending module includes using SIP-TLS transport SIP MESSAGE messages.
Furthermore, the sending module uses S/MIME mode to encrypt the text part separately.
Furthermore, the sending module adopts a timing sending mechanism, and the sender sends the characters converted from the voice at fixed time intervals.
Furthermore, the sending module adopts an intelligent sending mechanism, and the sending party determines the sending time and the sending length through the escape sentence-breaking function of the STT.
Further, the timing display mechanism comprises: the receiving party presets a timing length, displays the received characters in the time length in each time length, and locally displays the characters after accumulating a certain amount of character information by using a cache mechanism.
Furthermore, the receiving module adopts a timing display mechanism, and the receiving party displays the received characters at fixed intervals.
Furthermore, the receiving module displays the received characters on a local display screen in real time
Further, the S05 is to segment a SIP MESSAGE received text in a period and scroll the text on the display screen.
Furthermore, the text information is converted into a text and then stored locally, and a user manually turns pages to view the text information.
Further, the text is deleted in time after the call is completed.
The above-mentioned english abbreviation:
SIP: session Initiation Protocol, Session Initiation Protocol
VoIP: voice Internet Protocol
UAC: user Agent Client, User Agent Client
UAS: user Agent Server, User Agent Server
STT: speech to Text, Speech to Text
SIP MESSAGE: session Initiation Protocol MESSAGE
RFC 3428: session Initiation Protocol (SIP) extensions for Instant Messaging, Session Initiation Protocol (SIP) extensions for Instant Messaging
RTP: real-time Transport Protocol, Real-time Transport Protocol
NPU: neural-network Processing Unit, Neural network Processing Unit
SIP-TLS: session Initiation Protocol-Transport Layer Secure, Session Initiation Protocol-based Transport Layer security
S/MIME: Secure/Multipurpose Internet Mail Extensions for Secure/Multipurpos Internet Mail Extensions
The invention can enable the remote receiver to see the SIP text message synchronously displayed on the local display screen while receiving and playing the voice, only uses a small amount of extra bandwidth, enhances the understanding of the receiver to the voice, improves the communication quality, encrypts the text, and ensures the safety of the information and the protection of personal privacy.
It should be understood that the statements herein reciting aspects are not intended to limit the critical or essential features of any embodiment of the invention, nor are they intended to limit the scope of the invention. Other features of the present invention will become apparent from the following description.
Drawings
The above and other features, advantages and aspects of various embodiments of the present invention will become more apparent by referring to the following detailed description when taken in conjunction with the accompanying drawings. Wherein:
fig. 1 shows a flow chart of a method for improving VoIP voice call quality in accordance with an embodiment of the present invention;
fig. 2 is a block diagram illustrating an apparatus for improving VoIP voice call quality according to an embodiment of the present invention;
fig. 3 shows a schematic diagram according to an embodiment of the invention.
Detailed Description
In order to make the objects, technical solutions and advantages of the embodiments of the present invention clearer, the technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are some, but not all, embodiments of the present invention. All other embodiments, which can be obtained by a person skilled in the art without any inventive step based on the embodiments of the present invention, are within the scope of the present invention.
Alice and Bob use two UACs, the UAC has a physical display screen and supports a local STT function, the display screen can display 3 lines of characters at one time, each line has 15 characters, and the UAC can freely select to open or close the function of displaying the voice call characters.
And the UAC used by Alice sets a timing sending and receiving function, the set time duration is 5 seconds, namely the UAC sends or receives characters converted from voice within 5 seconds every 5 seconds, and sets a character segment received by SIP MESSAGE within 5 seconds to be displayed on a display screen in a rolling manner.
And setting a timed sending and receiving function by the UAC used by the Bob, wherein the set time duration is 5 seconds, namely, the UAC sends or receives the characters converted from the voice within 5 seconds, and sets manual opening and page turning to view character information.
The two UACs establish SIP voice conversation through the UAS, and respectively detect whether the voice transmitted to the local microphone by the local user is normal, if so, the voice conversation can be carried out.
Alice calls Bob, hello to Bob, this voice message is processed by STT analysis technique to complete the conversion from voice to text, and the text is sent SIP MESSAGE to UAS:
MESSAGE sip:bob@domain.com SIP/2.0
Via:SIP/2.0/TCP user1pc.domain.com；branch＝z9hG4bK776sgdkse
Max-Forwards:70
From:sip:alice@domain.com；tag＝49583
To:sip:bob@domain.com
Call-ID:asd88asd77a@1.2.3.4
CSeq:1MESSAGE
Content-Type:text/plain
Content-Length:11
Hello Bob.
the UAS sends SIP MESSAGE to Bob:
MESSAGE sip:bob@domain.com SIP/2.0
Via:SIP/2.0/TCP proxy.domain.com；branch＝z9hG4bK123dsghds
Via:SIP/2.0/TCP user1pc.domain.com；branch
＝z9hG4bK776sgdkse；received＝1.2.3.4
Max-Forwards:69
From:sip:alice@domain.com；tag＝49394
To:sip:bob@domain.com
Call-ID:asd88asd77a@1.2.3.4
CSeq:1MESSAGE
Content-Type:text/plain
Content-Length:11
Hello Bob.
the Bob end acknowledges the message 200OK, feeding back to the UAS:
SIP/2.0200OK
Via:SIP/2.0/TCP proxy.domain.com；branch
＝z9hG4bK123dsghds；received＝192.0.2.1
Via:SIP/2.0/TCP user1pc.domain.com；；branch＝
z9hG4bK776sgdkse；received＝1.2.3.4
From:sip:alice@domain.com；tag＝49394
To:sip:bob@domain.com；tag＝ab8asdasd9
Call-ID:asd88asd77a@1.2.3.4
CSeq:1MESSAGE
Content-Length:0
UAS acknowledge message 200OK, feedback to Alice end:
SIP/2.0200OK
Via:SIP/2.0/TCP proxy.domain.com；branch
＝z9hG4bK123dsghds；received＝192.0.2.1
Via:SIP/2.0/TCP user1pc.domain.com；；branch
＝z9hG4bK776sgdkse；received＝1.2.3.4
From:sip:alice@domain.com；tag＝49394
To:sip:bob@domain.com；tag＝ab8asdasd9
Call-ID:asd88asd77a@1.2.3.4
CSeq:1MESSAGE
Content-Length:0
and displaying 'Bob, hello' on the display screen of the Bob end, and hearing the voice of Alice.
In the process of message transmission, SIP MESSAGE messages are transmitted by using SIP-TLS, and the text part is separately encrypted by using an S/MIME mode, so that the safety of voice conversation is ensured. SIP MESSAGE message sending and receiving conform to the specification defined by RFC3428(Session Initiation Protocol (SIP) Extension for Instant Messaging), and is performed simultaneously with the existing RTP message interaction.
Alice continues to speak: "to receive notification, the company leader will take a meeting in the first meeting room of the company ten am on 2021 tomorrow, 8 months and 3 am. Please take part in time. When the time is 9 seconds, the UAC sends words every 5 seconds, the Bob end can check the text information sent by the UAC of the Alice end, if the text information needs to be checked by manually turning pages, the complete text information converted from the voice can be checked after 10 seconds, and the words are stored in the Bob end.
And (5) Bob replies: "what is the subject of the meeting? What are people participating? Which files need to be prepared? How long is the conference duration presumably? If the meal needs to be ordered beyond the meal point? When the time is 12 seconds, the UAC sends characters every 5 seconds, the display screen of the Alice terminal displays character information in a rolling mode, and the characters are displayed completely after 15 seconds. After the Bob end finishes the conversation, the UAC deletes the stored text.
Alice and Bob can check the text information converted by the voice while hearing the voice of the other party, and can complete the conversation through the text even under the condition of poor conversation quality, thereby improving the conversation quality.
Based on the same inventive concept, the invention also provides a device for improving the low quality of VoIP voice communication. The implementation of the device can be referred to the implementation of the method, and repeated details are not repeated. As shown in fig. 2, the apparatus 100 includes:
the connection module 101: the UAC is used for establishing SIP voice session through the UAS;
the detection module 102: the UAC is used for detecting the voice transmitted to a local microphone by a local user;
the voice conversion module 103: the UAC carries out character conversion on the detected voice through an STT analysis technology;
the sending module 104: for UAC to send text to UAS/remote user via SIP MESSAGE message;
the receiving module 105: the remote user can analyze the received SIP MESSAGE message and display the text information on a local display screen in real time.
The device for improving the low VoIP voice conversation quality can enable a far-end receiver to see the SIP text message synchronously displayed on the local display screen while receiving and playing the voice, only a small amount of extra bandwidth is used, the understanding of the receiver to the voice is enhanced, the conversation quality is improved, the text is encrypted, and the information safety and the personal privacy protection are ensured.
While the spirit and principles of the invention have been described with reference to several particular embodiments, it is to be understood that the invention is not limited to the disclosed embodiments, nor is the division of aspects, which is for convenience only as the features in such aspects may not be combined to benefit. The invention is intended to cover various modifications and equivalent arrangements included within the spirit and scope of the appended claims.
The limitation of the protection scope of the present invention is understood by those skilled in the art, and various modifications or changes which can be made by those skilled in the art without inventive efforts based on the technical solution of the present invention are still within the protection scope of the present invention.
Claims (22)
1. A method for improving VoIP voice call quality, the method comprising:
s01: the UAC establishes an SIP voice session through the UAS;
s02: the UAC detects the voice transmitted to a local microphone by a local user;
s03: the UAC performs character conversion on the detected voice through an STT analysis technology;
s04: the UAC sends the words to the UAS/remote user through SIPMSAGE message;
s05: and the remote user analyzes the received SIPMSAGE message and displays the text information on a local display screen.
2. The method as claimed in claim 1, wherein the UAC has a physical display.
3. The method as claimed in claim 2, wherein the physical display screen can display not less than 3 lines of text, each line displaying not less than 10 characters.
4. The method as claimed in claim 1, wherein the S04 transmits the SIP message using SIP-TLS.
5. The method as claimed in claim 1, wherein the S04 encrypts the text part separately using S/MIME.
6. The method as claimed in claim 1, wherein the S04 employs a timing transmission mechanism, and the transmitter transmits the text converted from the voice at regular intervals.
7. The method as claimed in claim 1, wherein the S04 is an intelligent transmission mechanism, and the sender determines the transmission timing and length by using an escape sentence-breaking function of STT.
8. The method as claimed in claim 1, wherein the S05 is a timing display mechanism, and the receiving party displays the received text at regular intervals.
9. The method as claimed in claim 1 or 7, wherein said S05 displays the received text on a local display screen in real time.
10. The method as claimed in claim 1, wherein the step S05 is performed by scrolling the received text segments on the display screen.
11. The method as claimed in claim 1, wherein the text message is converted into text and stored locally.
12. An apparatus for improving VoIP voice call quality, the apparatus comprising:
a connecting module: the UAC is used for establishing SIP voice session through the UAS;
a detection module: the UAC is used for detecting the voice transmitted to a local microphone by a local user;
a voice conversion module: the UAC carries out character conversion on the detected voice through an STT analysis technology;
a sending module: for UAC to send words to UAS/remote user through SIP MESSAGE message;
a receiving module: the remote user analyzes the received SIPMSAGE message and displays the text information on a local display screen.
13. The apparatus as recited in claim 13, wherein said UAC has a physical display.
14. An apparatus as claimed in claim 13, wherein the physical display screen can display no less than 3 lines of text, each line displaying no less than 10 characters.
15. The apparatus as claimed in claim 13, wherein the sending module transmits the SIP message using SIP-TLS.
16. The apparatus as claimed in claim 13, wherein the sending module encrypts the text portion separately using S/MIME.
17. The apparatus as claimed in claim 13, wherein the sending module employs a timing sending mechanism, and the sender sends the text converted from voice at regular intervals.
18. The method as claimed in claim 13, wherein the sending module employs an intelligent sending mechanism, and the sender determines the sending time and sending length by using an escape and sentence-break function of STT.
19. The apparatus as claimed in claim 13, wherein the receiving module employs a timing display mechanism, and the receiving party displays the received text at regular intervals.
20. An apparatus as claimed in claim 13 or 18, wherein the receiving module displays the received text on a local display screen in real time.
21. The apparatus as claimed in claim 13, wherein the receiving module is configured to segment the received text and scroll the segmented text on the display screen.
22. The apparatus as claimed in claim 13, wherein the text message is converted to text and stored locally.
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| US20170085506A1 (en) * | 2015-09-21 | 2017-03-23 | Beam Propulsion Lab Inc. | System and method of bidirectional transcripts for voice/text messaging |
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| CN101577767A (en) * | 2008-05-07 | 2009-11-11 | 阿尔卡特朗讯公司 | Real-time voice-to-text conversion for telecommunication services |
| CN102710539A (en) * | 2012-05-02 | 2012-10-03 | 中兴通讯股份有限公司 | Method and device for transferring voice messages |
| CN104639728A (en) * | 2013-11-14 | 2015-05-20 | 阿尔卡特朗讯公司 | Method and equipment used for improving speed or video communication quality |
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