CN103209442B - A kind of method and terminal that speech business configured transmission is set dynamically - Google Patents
A kind of method and terminal that speech business configured transmission is set dynamically Download PDFInfo
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Abstract
The present invention discloses a kind of method and terminal that speech business configured transmission is set dynamically.Methods described includes：Obtain the code/decode type of transmitting voice service；Configured transmission according to needed for the code/decode type sets transmission institute voice traffic.Using the method and terminal of the present invention, different code/decode types can be directed to, for this speech business, corresponding configured transmission is set, so as to avoid in the prior art as the waste to limited interface-free resources caused by different code/decode types distribution identical bandwidth, limited interface-free resources can be saved.
Description
Technical field
The present invention relates to the communications field, more particularly to a kind of method that speech business configured transmission is set dynamically and end
End.
Background technology
With worldwide interoperability for microwave accesses (Worldwide Interoperability for Microwave Access,
WiMAX) scale of network is commercial, and increasing user begins to use wireless network generally enter in the past in cable network
Capable business.For example, carrying out Video chat by wireless network, mass data etc. is downloaded.The bandwidth money that these business take
Source will be far longer than the bandwidth resources shared by common voice calling service.Especially in the wireless networks such as WiMAX, due to
Allow the interface-free resources (bandwidth resources i.e. in wireless network) that use limited, so more should rationally be utilized to bandwidth resources.
In the prior art, the terminal device of WiMAX network mostly has IP phone (Voice over
Internet Portocol, VOIP) function.VOIP is exactly to digitize simulated sound signal (Voice) in brief, with number
According to the form of package (Data Packet) real-time delivery is done on IP data networks (IPNetwork).In VOIP technologies, for
The code encoding/decoding mode of speech data has a variety of.Some code encoding/decoding modes are higher for the compression degree of speech data, take band
Wide resource is less, and speech quality is slightly worse；Some code encoding/decoding modes then for speech data compression degree than relatively low, occupied bandwidth
Resource is more, but speech quality is preferable.
The method of the prior art that speech business configured transmission is set, when terminal initiates dynamic service flow, network side
The parameters such as the bandwidth resources for terminal distribution are fixed.That is, no matter which kind of code encoding/decoding mode is reality use, all can be with a certain kind
It is terminal distribution band based on bandwidth resources required for code encoding/decoding mode (being typically the more mode of occupied bandwidth resource)
It is wide.Therefore, the bandwidth allocation methods of wireless network voice business of the prior art, wave is caused to limited interface-free resources
Take.
The content of the invention
, can be according to every it is an object of the invention to provide a kind of method and terminal that speech business configured transmission is set dynamically
The code encoding/decoding mode of secondary speech business, configured transmission required under this code encoding/decoding mode is determined, and then be each wireless network
Network speech business distributes corresponding bandwidth, can save limited interface-free resources.
To achieve the above object, the embodiments of the invention provide following scheme：
The embodiment of the present invention provides a kind of method that speech business configured transmission is set dynamically, including：Obtain transmission voice
The code/decode type of business；Configured transmission according to needed for the code/decode type sets transmission institute voice traffic.
The embodiment of the present invention also provides a kind of terminal, including：Code/decode type acquiring unit, voice industry is transmitted for obtaining
The code/decode type of business；Transmission parameter settings unit, for setting voice traffic institute of transmission institute according to the code/decode type
The configured transmission needed.
According to specific embodiment provided by the invention, the embodiment of the present invention realizes following technique effect：
The method of dynamic setting speech business configured transmission disclosed in this invention, it is contemplated that right under different code/decode types
This different factor of configured transmission answered, by the code/decode type for obtaining transmitting voice service；According to the code/decode type
Configured transmission needed for transmission institute voice traffic is set；Different code/decode types can be directed to, are set for this speech business
Corresponding configured transmission, so as to avoid in the prior art for different code/decode types distribute identical bandwidth caused by having
The waste of the interface-free resources of limit, limited interface-free resources can be saved.
Brief description of the drawings
In order to illustrate more clearly about the embodiment of the present invention or technical scheme of the prior art, below will be to institute in embodiment
The accompanying drawing needed to use is briefly described, it should be apparent that, drawings in the following description are only some implementations of the present invention
Example, for those of ordinary skill in the art, without having to pay creative labor, can also be according to these accompanying drawings
Obtain other accompanying drawings.
The method flow diagram for the dynamic setting speech business configured transmission that Fig. 1 is provided by the embodiment of the present invention one；
The method flow diagram for the dynamic setting speech business configured transmission that Fig. 2 is provided by the embodiment of the present invention two；
The structure chart for the terminal that Fig. 3 is provided by the embodiment of the present invention three；
The structure chart for the terminal that Fig. 4 is provided by the embodiment of the present invention four.
Embodiment
Below in conjunction with the accompanying drawing in the embodiment of the present invention, the technical scheme in the embodiment of the present invention is carried out clear, complete
Site preparation describes, it is clear that described embodiment is only part of the embodiment of the present invention, rather than whole embodiments.It is based on
Embodiment in the present invention, those of ordinary skill in the art are obtained every other under the premise of creative work is not made
Embodiment, belong to the scope of protection of the invention.
In order to facilitate the understanding of the purposes, features and advantages of the present invention, it is below in conjunction with the accompanying drawings and specific real
Applying mode, the present invention is further detailed explanation.
Embodiment one
The method flow diagram for the dynamic setting speech business configured transmission that Fig. 1 is provided by the embodiment of the present invention one.This reality
The executive agent for applying example can be a kind of wireless networking terminal device.As shown in figure 1, the method comprising the steps of：
S101：Obtain the code/decode type of transmitting voice service；
In practical application, code encoding/decoding mode can be consulted with the server of network side by initiating the terminal of business.When terminal with
After server has consulted code/decode type, negotiation result is sent to server, optionally, terminal can pass through negotiation result
THP message (Tomlinson-Harashima Precoding, Tomlinson-Harrar wish agate precoding) passes to WiMAX sides
Server.The server of WiMAX sides can be scheduled according to negotiation result to this speech business.
Specifically, when consulting code/decode type, the code encoding/decoding mode or current hair that can be supported according to the terminal
Requirement of the business risen to signal quality, selects suitable code encoding/decoding mode.For example, when terminal supports silent pressure with server
During contracting function, the code encoding/decoding mode with silence compression function can be selected；When this business is higher to demand on signal quality,
The code encoding/decoding mode that occupied bandwidth is more but signal quality is high can be selected.
S102：Configured transmission according to needed for the code/decode type sets transmission institute voice traffic.
Configured transmission according to needed for the code/decode type sets transmission institute voice traffic, including：According to the volume
Peak transfer rate, minimum transmission rate, propagation delay time, shake needed for coding type setting transmission institute voice traffic etc. pass
Defeated parameter.
Wherein, the configured transmission such as propagation delay time, shake is network quality etc. corresponding to the bandwidth as needed for this speech business
What level was determined.Optionally, the parameter such as propagation delay time, shake corresponding to the network quality levels, is referred to following table, following
Form comes from communication industry standard YD/T 1071-2000《The requirement of IP phone gateway equipment and technology》.
| Network quality levels | One Way Delay (ms) | Loss rate | Shake (ms) |
| Well (self-defined) | ≤40 | ≈0 | ≤10 |
| It is poor | ≤100 | ≤ 1% | ≤20 |
| Badly | ≤400 | ≤ 5% | ≤60 |
In WiMAX system, code encoding/decoding mode can use G.711, G.729, the encoding and decoding technique such as G.723.Specifically,
G.711 pulse code modulation (Pulse-code modulation, PCM) is also referred to as, be that International Telecommunication Union stipulates out one
Voice compression is covered, is mainly used in phone., mainly with pulse code modulation to audio sample, sample rate is that 8k is per second for it.It
Utilize the uncompressed channel transfer speech sound signals of a 64Kbps.It is 1: 2 to play compression ratio, i.e., 16 data is compressed into 8.
G.723.1 it is that International Telecommunication Union's telecommunication standardsization tissue worked out a kind of multi-media voice encoding and decoding being molded in 1996
Standard.Its typical case includes VoIP service, visual telephone, radio telephone, digital satellite system, the electric multiplying device of number, public
Switched telephone and various multi-media voice information products.G.723.1 standard transmission code check has 5.3kb/s and 6.3kb/s two
Kind.G.729 encoding scheme is the standard of the voice signal coding of telephone bandwidth, and the analog signal for inputting speech naturalness is used
8kHz, sampling, 16 bit linear PCM quantizations.
It can be seen that under different code encoding/decoding modes, the compression degree for speech data is different, and therefore, difference compiles solution
The actually required bandwidth of code mode also differs.So according to needed for the code/decode type sets transmission institute voice traffic
Configured transmission include：Bandwidth according to needed for the code/decode type calculates transmission institute voice traffic.
Every kind of code encoding/decoding mode all defines corresponding encoding and decoding standard.Wherein, including packing the cycle and it is per second packing length
Degree.Because speech data is typically the transmission in the form of packet, the packing cycle means that each packet institute of encapsulation
The time needed, unit is usually Millisecond.Packing length per second is exactly the having of being included of all packets of encapsulation in each second
Imitate load data amount.
Data packaged by each packet can be largely classified into two parts.A part is fixed field, and another part is
Payload.The fixed field is primarily referred to as the protocol fields that each packet carries, such as：Real time transport protocol field,
User Datagram Protocol field, procotol field and Ethernet field etc..Because the length of these fields compiles solution every kind of
All it is fixed in code mode, therefore the field length shared by these fields is also known, referred to as fixed field.Payload
Refer to the part data for representing voice messaging.In different code encoding/decoding modes, the payload in each packet is not yet
It is identical.Payload=packing cycle (in seconds) × packing per second length.
Therefore, can according to the packing length per second and the fixed field length of pack cycle and each packet,
Calculate the bandwidth value needed for the wireless network voice business；The fixed field length is the agreement word that each packet carries
The field length of section；
Specifically, G.711, G.729, G.723.1 (5.3kbit/s), the G.723.1 coded system such as (6.3Kbit/s)
In, the length of fixed field can calculate.Generally, fixed field mainly includes：Real time transport protocol field, user data
Packet protocol field, procotol field and Ethernet field.Wherein, real time transport protocol field (Real-time Transport
Protocol, RTP) length be 96bit, the length of User Datagram Protocol field (User Datagram Protocol, UDP)
Spend for 64bit, the length of procotol field (Internet Protocol, IP) is 160bit, Ethernet (Ethernet) word
The length of section is 208bit.The length summation of aforementioned four field, it is 528bit.So the length of fixed field can be used
528bit is represented.
The following detailed description of how according to the packing length per second and the fixed word of pack cycle and each packet
Segment length, calculate the bandwidth value needed for the wireless network voice business.
Required bandwidth=individual data packet length × packing per second number.Wherein individual data packet length=fixed field length
+ payload, packing number=1/ per second are packed the cycle.Payload=packing cycle (in seconds) × packing per second length
Degree.Therefore, after the known packing length per second and the fixed field length of packing cycle and each packet, Ke Yiyong
The packing length per second is multiplied by the packing cycle, obtains payload, and the payload is to represent voice in packet
The data of information；It is added with the payload with the fixed field length, obtains individual data packet length；Beaten according to described
The bag cycle converts to obtain packing number per second；The individual data packet length is multiplied by with the packing number per second, obtains the bandwidth
Value.
It can be seen that the method for dynamic setting speech business configured transmission disclosed in this invention, it is contemplated that different encoding and decoding classes
This different factor of corresponding configured transmission under type, by the code/decode type for obtaining transmitting voice service；Solved according to described compile
Code type sets the configured transmission needed for transmission institute voice traffic；Different code/decode types can be directed to, are this voice industry
Business sets corresponding configured transmission, is distributed in the prior art for different code/decode types caused by identical bandwidth so as to avoid
The waste to limited interface-free resources, limited interface-free resources can be saved.
Embodiment two
The method flow diagram for the dynamic setting speech business configured transmission that Fig. 2 is provided by the embodiment of the present invention two.Such as Fig. 2
Shown, the method comprising the steps of：
S201：Obtain the code/decode type of transmitting voice service；
S202：Bandwidth according to needed for the code/decode type calculates transmission institute voice traffic；
S203：According to network quality levels corresponding to the required bandwidth, the biography needed for transmission institute voice traffic is set
Defeated parameter；
S204：Obtain whether institute's voice traffic supports silence compression function；
In voice call process, it is possible that the pause (such as due to silence caused by thinking etc.) of of short duration time.
So, when of short duration pause occurs in a side of call, due to the characteristic of speech business, in the situation of no any voice messaging
Under, it is desired nonetheless to the side's transmission information for answering call, the information is used to represent that now speaker not to send any voice.
Silence compression just refers to, when caller does not send voice, represents that caller is in silent status with a mark, only needs to send out
The mark is given, the background sound without retransmiting caller's local environment.Because the data volume of mark is substantially smaller in size than background sound
The data volume size of information, so silence compression function can save bandwidth resources.
S205：When institute's voice traffic supports silence compression function, real-time wheel of the scheduling type for extension is set
Inquiry business ERTPS；
The real-time polling service (Extended Rt-Polling, ERTPS) of extension is that a kind of support of the prior art is quiet
The dispatching method of silent compression function.
S206：When institute's voice traffic does not support silence compression function, the scheduling type is set actively to authorize industry
Be engaged in UGS.
Active grant bandwidth business (Unsolicited Grant Service, UGS) is that one kind of the prior art is not propped up
Hold the dispatching method of silence compression function.
The present embodiment, different code/decode types can also be directed to, corresponding configured transmission is set for this speech business, from
And avoid the wave to limited interface-free resources distributed in the prior art for different code/decode types caused by identical bandwidth
Take, limited interface-free resources can be saved.
Whether the present embodiment is propped up compared with a upper embodiment by the terminal for judging to be related to the wireless network voice business
The dispatching method with silence compression function is held, when terminal supports silence compression, using the tune of the real-time polling service of extension
Degree method, when terminal does not support silence compression, using the dispatching method of active grant bandwidth business to the wireless network language
Sound business is scheduled, and can also further save the interface-free resources of wireless network.
Embodiment three
The structure chart for the terminal that Fig. 3 is provided by the embodiment of the present invention three.As shown in figure 3, the terminal includes：
Code/decode type acquiring unit 301, for obtaining the code/decode type of transmitting voice service；
Transmission parameter settings unit 302, needed for transmission institute voice traffic is set according to the code/decode type
Configured transmission.
It can be seen that terminal disclosed in this invention, it is contemplated that under different code/decode types corresponding configured transmission it is different this
Factor, by the code/decode type for obtaining transmitting voice service；Transmission institute voice traffic is set according to the code/decode type
Required configured transmission；Different code/decode types can be directed to, corresponding configured transmission are set for this speech business, so as to keep away
The waste to limited interface-free resources distributed in the prior art for different code/decode types caused by identical bandwidth, energy are exempted from
Enough save limited interface-free resources.
Example IV
The structure chart for the terminal that Fig. 4 is provided by the embodiment of the present invention four.As shown in figure 4, the terminal includes：
Code/decode type acquiring unit 401, for obtaining the code/decode type of transmitting voice service；
Transmission parameter settings unit 402, the configured transmission include：When peak transfer rate, minimum transmission rate, transmission
Prolong, shake, one or more of the network parameter such as scheduling type.
The transmission parameter settings unit 402 can include：
Transmission rate sets subelement 4021, needed for determining transmission institute voice traffic according to the code/decode type
Peak transfer rate and minimum transmission rate.
Bandwidth calculation subelement 4022, for calculating the band needed for transmitting institute's voice traffic according to the code/decode type
It is wide；
Other transmission parameter settings subelements 4023, for the network quality levels according to corresponding to the required bandwidth, if
Put the propagation delay time needed for transmission institute voice traffic, shake.
Silence compression function supports information acquisition unit 403, and for obtaining, whether voice traffic supports silence compression
Function；
Scheduling type setting unit 404, for whether supporting silence compression function setting to transmit according to institute's voice traffic
Scheduling type needed for institute's voice traffic；
The scheduling type setting unit 404 can include：
Real-time polling service scheduling type sets subelement 4041, for supporting silence compression function when institute's voice traffic
When, real-time polling service ERTPS of the scheduling type for extension is set；
Unsolicited Grant Service scheduling type sets subelement 4042, for not supporting silence compression work(when institute's voice traffic
During energy, it is Unsolicited Grant Service UGS to set the scheduling type.
Whether the present embodiment is propped up compared with a upper embodiment by the terminal for judging to be related to the wireless network voice business
The dispatching method with silence compression function is held, when terminal supports silence compression, using the tune of the real-time polling service of extension
Degree method, when terminal does not support silence compression, using the dispatching method of active grant bandwidth business to the wireless network language
Sound business is scheduled, and can also further save the interface-free resources of wireless network.
Each embodiment is described by the way of progressive in this specification, what each embodiment stressed be and other
The difference of embodiment, between each embodiment identical similar portion mutually referring to.For terminal disclosed in embodiment
For, because it is corresponded to the method disclosed in Example, so description is fairly simple, related part is said referring to method part
It is bright.
Professional further appreciates that, with reference to the unit of each example of the embodiments described herein description
And algorithm steps, it can be realized by electronic hardware or electronic hardware in a manner of computer software is combined.In order to clear
The interchangeability of ground declaratives hardware and software, generally describes each example according to function in the above description
Composition and step.These functions are performed in a manner of the hardware or software and hardware combining actually, specific depending on technical scheme
Using and design constraint.Professional and technical personnel can be described to be realized using distinct methods to each specific application
Function, but it is this realization it is not considered that beyond the scope of this invention.
Directly it can be held with reference to the step of method or algorithm that the embodiments described herein describes with hardware, processor
Capable software module, or the two combination are implemented.Software module can be placed in random access memory (RAM), internal memory, read-only deposit
Reservoir (ROM), electrically programmable ROM, electrically erasable ROM, register, hard disk, moveable magnetic disc, CD-ROM or technology
In any other form of storage medium well known in field.
The foregoing description of the disclosed embodiments, professional and technical personnel in the field are enable to realize or using the present invention.
A variety of modifications to these embodiments will be apparent for those skilled in the art, as defined herein
General Principle can be realized in other embodiments without departing from the spirit or scope of the present invention.Therefore, it is of the invention
The embodiments shown herein is not intended to be limited to, and is to fit to and principles disclosed herein and features of novelty phase one
The most wide scope caused.
Claims (2)
- A kind of 1. method that speech business configured transmission is set dynamically, it is characterised in that methods described includes：Obtain the code/decode type of transmitting voice service；Bandwidth according to needed for the code/decode type calculates transmission institute voice traffic；According to network quality levels corresponding to the required bandwidth, the configured transmission needed for transmission institute voice traffic is set；Obtain whether institute's voice traffic supports silence compression function；Scheduling type according to needed for whether institute's voice traffic supports silence compression function setting transmission institute voice traffic；Wherein, the bandwidth according to needed for the code/decode type calculates transmission institute voice traffic includes：According to packing length per second and the fixed field length computation transmission voice industry of pack cycle and each packet Bandwidth needed for business；The fixed field length is the field length for the protocol fields that each packet carries；The configured transmission includes：One in peak transfer rate, minimum transmission rate, propagation delay time, shake, scheduling type It is or multiple；The scheduling class according to needed for whether institute's voice traffic supports silence compression function setting transmission institute voice traffic Type includes：When institute's voice traffic supports silence compression function, real-time polling service of the scheduling type for extension is set ERTPS；When institute's voice traffic does not support silence compression function, it is Unsolicited Grant Service UGS to set the scheduling type.
- A kind of 2. terminal, it is characterised in that including：Code/decode type acquiring unit, for obtaining the code/decode type of transmitting voice service；Transmission parameter settings unit, join for the transmission needed for setting transmission institute voice traffic according to the code/decode type Number；Silence compression function supports information acquisition unit, and for obtaining, whether voice traffic supports silence compression function；Scheduling type setting unit, for whether supporting silence compression function setting to transmit the voice according to institute's voice traffic Scheduling type needed for business；The transmission parameter settings unit includes：Bandwidth calculation subelement, for calculating the bandwidth needed for transmitting institute's voice traffic according to the code/decode type；Transmission parameter settings subelement, for the network quality levels according to corresponding to the required bandwidth, transmission institute predicate is set Configured transmission needed for sound business；Wherein, the bandwidth calculation subelement is specifically used for according to packing length per second and pack cycle and each packet Bandwidth needed for fixed field length computation transmission institute voice traffic；The fixed field length is what each packet carried The field length of protocol fields；The configured transmission includes：One in peak transfer rate, minimum transmission rate, propagation delay time, shake, scheduling type It is or multiple；The scheduling type setting unit includes：Real-time polling service scheduling type sets subelement, for when institute's voice traffic supports silence compression function, setting The scheduling type is the real-time polling service ERTPS of extension；Unsolicited Grant Service scheduling type sets subelement, for when institute's voice traffic does not support silence compression function, if It is Unsolicited Grant Service UGS to put the scheduling type.
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| CN106657638A (en) * | 2016-12-23 | 2017-05-10 | 宇龙计算机通信科技(深圳)有限公司 | Communication method and communication device based on call content, and terminal |
| CN109729552B (en) * | 2017-10-27 | 2022-03-25 | 成都鼎桥通信技术有限公司 | Voice transmission method and device |
| CN112511782B (en) * | 2019-09-16 | 2024-05-07 | 中兴通讯股份有限公司 | Video conference method, first terminal, MCU, system and storage medium |
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| CN1486076A (en) * | 2002-09-26 | 2004-03-31 | 华为技术有限公司 | Method for dynamic control of voice bandwidth |
| CN101056466A (en) * | 2007-02-12 | 2007-10-17 | 华为技术有限公司 | A method and device for adjusting the voice encoding and decoding mode in the call process |
| CN101820686A (en) * | 2010-04-29 | 2010-09-01 | 京信通信系统（中国）有限公司 | Uplink bandwidth allocation method and system for WiMAX system |
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| CA2379412C (en) * | 1999-07-19 | 2007-06-05 | At&T Corp. | A multiple-access scheme for packet voice that uses voice activity detection |
| FI115329B (en) * | 2000-05-08 | 2005-04-15 | Nokia Corp | Method and arrangement for switching the source signal bandwidth in a communication connection equipped for many bandwidths |
| KR20090030562A (en) * | 2007-09-20 | 2009-03-25 | 엘지전자 주식회사 | Method of packet transmission for resource allocation in broadband wireless access system |
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| CN1486076A (en) * | 2002-09-26 | 2004-03-31 | 华为技术有限公司 | Method for dynamic control of voice bandwidth |
| CN101056466A (en) * | 2007-02-12 | 2007-10-17 | 华为技术有限公司 | A method and device for adjusting the voice encoding and decoding mode in the call process |
| CN101820686A (en) * | 2010-04-29 | 2010-09-01 | 京信通信系统（中国）有限公司 | Uplink bandwidth allocation method and system for WiMAX system |
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Effective date of registration: 20181218 Address after: 523808 Southern Factory Building (Phase I) Project B2 Production Plant-5, New Town Avenue, Songshan Lake High-tech Industrial Development Zone, Dongguan City, Guangdong Province Patentee after: HUAWEI DEVICE Co.,Ltd. Address before: 518129 Building 2, B District, Bantian HUAWEI base, Longgang District, Shenzhen, Guangdong. Patentee before: Huawei terminal (Shenzhen) Co.,Ltd. |
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| TR01 | Transfer of patent right | ||
| TR01 | Transfer of patent right |
Effective date of registration: 20210420 Address after: Unit 3401, unit a, building 6, Shenye Zhongcheng, No. 8089, Hongli West Road, Donghai community, Xiangmihu street, Futian District, Shenzhen, Guangdong 518040 Patentee after: Honor Device Co.,Ltd. Address before: Metro Songshan Lake high tech Industrial Development Zone, Guangdong Province, Dongguan City Road 523808 No. 2 South Factory (1) project B2 -5 production workshop Patentee before: HUAWEI DEVICE Co.,Ltd. |