CN100484100C - Distributing method for VOIP service band width - Google Patents
Distributing method for VOIP service band width Download PDFInfo
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- CN100484100C CN100484100C CNB2006100616841A CN200610061684A CN100484100C CN 100484100 C CN100484100 C CN 100484100C CN B2006100616841 A CNB2006100616841 A CN B2006100616841A CN 200610061684 A CN200610061684 A CN 200610061684A CN 100484100 C CN100484100 C CN 100484100C
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Abstract
Thisinvention relates to a distribution method for VOIP service bandwidth including: setting bandwidth classes of tansmission channels according to the maximum rate and ensured rate of the VOIP service assigned by RAB, dynamically adjusting the clases of the VOIP service bandwidth by monitoring the load resource of the transmission channels to distribute a large bandwidth when the local resource is normal to secure the high phone quality and distribute the bandwidth based on the ensuring rate when the local resource is congested to reduce the local congestion so as to increase the usability of the bandwidth of the transmission channels and ensure the QoS of other services of users as well.
Description
Technical field
The present invention relates to the communication technology, relate in particular to the bandwidth management techniques in the communications field.
Background technology
Traditional telephone network is with the circuit exchange mode transferring voice, desired transmission broadband is 64kbit/s. and because telephone service always is the strictest business of various countries' control, and international long-distance telephone expense in various countries' exists serious disequilibrium, the international long-distance telephone business is all monopolized in a lot of countries, so, along with the development of internet Internet, realize that on Internet voice call becomes a kind of trend.
Originally, utilize the speech business on the software realization Internet, the user only needs to install client software on PC, and cooperate equipment such as microphone, sound card, sound equipment, just can on the IP network with the same user's communication that these software and hardwares are installed. used owing to be only limited on Internet at that time, therefore be commonly referred to ＂ Internet phone ＂, i.e. IP (Internet Protocol) phone.
Development along with technology, realize Internet and existing public switched telephone network (PSTN) (PSTN, PublicSwitched Telephone Network) combination, make IP phone develop into from originally PC to PC PC to PC, PC to phone, phone is to multiple business forms such as phones, and to the transition of IP transmitting multimedia service. but come what may, the IP phone bearer network is Internet, or follows the private network or the Internet. of ICP/IP protocol
As at UMTS (Universal Mobile Telecommunications System, universal mobile telecommunications system) in, UMTS is the 3-G (Generation Three mobile communication system) that adopts WCDMA (Wideband Code Division Multiple Access, Wideband Code Division Multiple Access (WCDMA) inserts) air interface technologies.
As shown in Figure 1, the UMTS system comprises RAN (Radio Access Network, wireless access network) and CN (Core Network, core net). wherein RAN is used to handle all and wireless function associated, all voice calls are connected with data in the UMTS system and CN handles, and the exchange of realization and external network and routing function .CN are from being divided into CS (Circuit Switched Domain in logic, circuit commutative field) and PS (Packet Switched Domain, packet-switched domain), CS generally comprises MSC (MobileSwitching Center, mobile switching centre)/VLR (Visitor Location Register, VLR Visitor Location Register), GMSC (Gateway Mobile Switching Center, Gateway Mobile Switching Center), gsmSSF, PS generally comprises SGSN (Serving GPRS Support Node, Serving GPRS Support Node), GGSN (Gateway GPRS Support Node, Gateway GPRS Support Node); CS mainly handles speech businesses such as relevant phone, voice, PS then handles relevant Packet data service .UTRAN (UMTS Territorial Radio Access Network UMTS, land radio access web), CN and UE (User Equipment, user terminal) constituted whole UMTS system together, the UMTS system connects external network, for example: PSTN and the Internet.
Above-mentioned land radio access web UTRAN, its network configuration block diagram as shown in Figure 2, comprise at least one RNS (Radio Network Subsystem, RNS), a RNS is by a RNC (Radio Network Controller, radio network controller) and at least one NodeB (base station) form, NodeB can cover at least one sub-district CELL.
At present, UTRAN uses the Iu series interfaces, comprises Iu, Iur and Iub interface.
Interface between RNC and the CN is the Iu interface, and NodeB is connected by Iub interface with RNC, and the user terminal UE among NodeB and its sub-district CELL is by the Uu interface communication.The Iu interface section of the circuit commutative field CS of core net CN is called Iu_CS, the Iu interface section of packet-switched domain PS is called Iu_PS. in UTRAN inside, interconnected between the RNC by Iur, Iur can connect or connect by transmission network by the direct physical between the RNC, RNC is used for distributing and controls and is attached thereto or the Radio Resource .NodeB of relevant NodeB then finishes the conversion of the data flow between Iub interface and the Uu interface, simultaneously also participate in a part of RRM, wherein:
NodeB is the base station (being radio receiving-transmitting unit) of WCDMA system, comprise radio receiving-transmitting unit and Base-Band Processing parts. by the Iub interface and the RNC interconnection of standard, mainly finish the processing of Uu interface physical layer protocol, major function is spread spectrum, modulation, chnnel coding and despreading, demodulation, channel-decoding, also comprises the functions such as mutual conversion of baseband signal and radiofrequency signal.
RNC is a radio network controller, is used to control the Radio Resource of UTRAN, mainly finishes functions such as connection foundation and disconnection, switching, the merging of grand diversity, RRM control.
The Iu interface is divided into Iu control plane and Iu user plane again, the Iu control plane transmits signaling, the Iu user plane transmits the transmission of user data .Iu interface and adopts ATM, Iu adopts AAL2 (the ATM Adaptation Layer type 2 of ATM at user plane, AAL 2) adapting protocol carrying user face data, the professional main PVC (Permanent Virtual Circuit, permanent virtual circuit) of ATM that uses is as carrying.
Iub interface between RNC and NodeB generally uses a plurality of AAL2 PVC to carry the data of UE, and these data comprise CS voice, the PS data of UE.
Carrying out voice communication by Internet is a very complicated system engineering, and related technology is also more, and wherein the most basic technology is packet voice (VoIP, a Voice over IP) technology.
VoIP is to be transmission platform with the IP packet switching network, seeing through the speech sound signal of IP network transmission or the technology of image signal. it is by a series of transcoding, coding, compression, the packing supervisor, so that speech data can be transferred to destination on IP network, and then via opposite program, being reduced into original speech sound signal receives .VOIP for those who answer and is based upon the technical packetizing of IP, the digitlization transmission technology, its basic principle is: by voice compression algorithm speech data is carried out compressed encoding and handle, then these speech datas are packed by related protocols such as IP, through IP network data packet transmission is arrived reception ground, again these VoPs are stringed together, after the decoding decompression processing, revert to original voice signal, thereby the purpose .IP telephone system that reaches by IP network transmission voice becomes the analog signal conversion of plain old telephone computer can link the IP packet that the internet transmits, also the IP packet of receiving being converted simultaneously to the analog electrical signal of sound. conversion and compression through IP telephony system are handled, each plain old telephone transmission rate takies 8～11Kbit/s bandwidth approximately, therefore when using transmission rate as the bandwidth of 64kbit/s equally with common telecommunications network, the IP phone number is original 5～8 times.
At present at Wideband Code Division Multiple Access (WCDMA) (WCDMA, Wideband Code Division Multiple Access) in the system, voice adopt adaptive multi-rate (AMR, Adaptive Multi-Rate) compressed encoding, being converted to the IP packet then, to transmit .AMR coding on IP network be a kind of adaptive coding method, can produce 8 kinds of different patterns, each pattern is corresponding to a kind of speed: 12.2,10.2,7.95,7.4,6.7,5.9,5.15 and 4.75kbit/s. is at bLock error rate (BLER, Block error rate) smaller or equal under 1% the condition, pattern is high more, the voice quality that provides is high more, but the transmission channel bandwidth resource (comprising load resource and Iub resource) that takies is also many more.
Carrying out RAB (RAB, Radio Access Bearer) when setting up, at first send the RAB assignment request message to UTRAN by CN, request UTRAN sets up RAB, the core net CN meeting corresponding service quality of assignment (QoS, Quality of Service) parameter, radio network controller (RNC, RadioNetwork Controller) according to the QoS of different mode, is VOIP traffic assignments corresponding bandwidth resource.
At present, RNC does not generally consider transmission channel resources, according to the maximum rate mode assignments bandwidth resources of RAB assignment, as the maximum rate when the RAB assignment is 12.2K, when guaranteeing that speed is 10.2K, RNC distributes the transmission channel bandwidth of 12.2K to be used for voice communication according to maximum rate 12.2K.Present this transmission channel resources distribution method, when local resource is sufficient, distribute bandwidth that better voice quality can be provided according to maximum rate, but when local resource is congested, when distributing bandwidth to be used for voice communication according to maximum rate, might increase the weight of the congested of sub-district, cause user's partial service impaired easily, influence QoS.Simultaneously, the actual bandwidth that needs when speech data more after a little while, waste bandwidth resource greatly.
Summary of the invention
In view of this, the invention provides a kind of distribution method of VOIP service bandwidth, guarantee that other business of user has better QoS simultaneously, can improve the utilance of transmission channel bandwidth again.
A kind of distribution method of VOIP service bandwidth wherein, comprising:
Steps A is according to the maximum rate of the VOIP business of RAB assignment with guarantee that speed is provided with the transmission channel bandwidth grade;
Step B, detected transmission channel loading situation;
Step C adjusts the grade of VOIP service bandwidth according to detected transmission channel load, when described transmission channel load surpasses default thresholding, step by step or skip a grade and reduce the grade of described VOIP service bandwidth; And/or, when described transmission channel load is lower than default thresholding, then rise step by step or the grade of the high described VOIP service bandwidth of skipping a grade.
Compared with prior art, the distribution method of VOIP service bandwidth of the present invention, because in advance according to the maximum rate of the VOIP business of RAB assignment with guarantee that speed is provided with the transmission channel bandwidth grade, then can dynamically adjust the grade of VOIP service bandwidth by monitoring transmission channel load resource situation, then when local resource is normal, distribute big bandwidth, guarantee high voice quality; When local resource is congested,,, thus, guarantee that other business of user has better QoS simultaneously, can improve the utilance of transmission channel bandwidth again to alleviate the small area jam situation according to guaranteeing the rate-allocation bandwidth.
Description of drawings
Fig. 1 is the network configuration block diagram of the UMTS system of prior art.
Fig. 2 is the network configuration block diagram of the UTRAN of prior art.
Fig. 3 is the method flow block diagram of the present invention's better embodiment.
Embodiment
For making the purpose, technical solutions and advantages of the present invention clearer, below in conjunction with embodiment and accompanying drawing, the present invention is further detailed explanation.
The distribution method of a kind of VOIP service bandwidth of the present invention mainly is dynamically to distribute bandwidth according to the QOS and the local resource situation of VOIP business, distributes big bandwidth when local resource normal (all not congested as load, Iub resource), guarantees high voice quality; When local resource congested (as load or Iub resource congestion) according to guaranteeing the rate-allocation bandwidth, to alleviate the small area jam situation.
As shown in Figure 3, the distribution method FB(flow block) for a kind of VOIP service bandwidth of the present invention's better embodiment mainly comprises the steps.
At first, RNC is provided with Iub load-threshold between cell load thresholding, RNC and its affiliated each NodeB, and RNC can also further be provided with the resource statistics cycle.
Carrying out RAB RAB when setting up, at first sending the RAB assignment request message by CN to UTRAN, request UTRAN sets up RAB, the corresponding service quality of core net CN assignment (QoS, Quality of Service) parameter.
According to corresponding service quality (QoS, Quality of Service) parameter, RAB sets up in the process, the maximum rate of assignment VOIP business and assurance speed in the RAB assignment signaling, maximum rate is meant the maximum rate that transferring voice reached between UE and the CN, at this moment, voice quality is the highest, the then corresponding bandwidth that can require maximum; Guarantee that speed is meant under the prerequisite that will guarantee the QoS that sets, between UE and the CN transferring voice the speed that must reach.
RNC is according to the maximum rate and the assurance speed division transmission channel bandwidth grade of RAB assignment, and generally speaking, the corresponding maximum rate of maximum bandwidth guarantees the corresponding assurance of bandwidth speed, and the intergrade bandwidth is selected successively according to the speed between maximum rate and the assurance speed.
For example, when the maximum rate of RAB assignment is 12.2K, when guaranteeing that speed is 7.95K, the transmission channel bandwidth grade can be divided into 3 grades, is followed successively by: 12.2K corresponding transmission channel bandwidth, 10.2K corresponding transmission channel bandwidth, 7.95K corresponding transmission channel bandwidth.
Certainly, as required, can be a plurality of ranks according to the maximum rate and the assurance speed division transmission channel bandwidth grade of RAB assignment, as at least three ranks.
The transmission channel load mainly comprises based on the cell load resource of power with based on the IUB load resource, so RNC detected cells load in real time and/or Iub load, or add up the throughput of sub-district and/or the throughput of Iub in the cycle, and then obtain cell load and/or Iub load at resource statistics.
In the present embodiment, judge whether the transmission channel load surpasses default load-threshold by judging whether whether cell load surpass the realization of Iub load-threshold above load-threshold and/or Iub load.
Detected cells load in real time and/or Iub load, or add up in the cycle at resource statistics and to obtain cell load and/or Iub load, judge whether cell load surpasses load-threshold and/or whether the Iub load surpasses the Iub load-threshold, and it is congested to judge promptly whether sub-district and/or Iub resource all take place.
If it is congested that sub-district and Iub resource all do not take place, then proceed the transmission of voice, data and multimedia service, and execution in step 103, step 104 again, up to judge sub-district or Iub resource take place one of at least congested, execution in step 105.
Carry out in the process in business, one of at least take place when congested when judging sub-district or Iub resource, then triggering the VOIP business carries out transmission channel and reshuffles, on existing bandwidth basis, reduce the VOIP bandwidth class step by step, also can be according to the load-threshold reduction bandwidth class of skipping a grade of configuration, up to the VOIP bandwidth class is reduced to the assurance rate bandwidth.
Through after the transfer of data of certain hour, possible sub-district and Iub resource all do not take place congested, monitor promptly that cell load is lower than its load-threshold and the Iub load is lower than its load-threshold, VOIP bandwidth class step by step can raise on existing bandwidth basis, also can be according to the load-threshold rising VOIP bandwidth class of skipping a grade of configuration, until being raised to maximum bandwidth.
In addition, can also configuration cycle timer, timer expiry judges whether the transmission channel load is higher or lower than default load-threshold, if then carry out above-mentioned corresponding operation; Timer is overtime more next time, judges whether load is higher or lower than thresholding again.Be specially: timer expiry, judge when cell load and Iub load all do not exceed separately load-threshold, VOIP bandwidth class step by step then raises on existing bandwidth basis, also can be according to the load-threshold rising VOIP bandwidth class of skipping a grade of configuration, until being raised to maximum bandwidth, close cycle timer; Judge when cell load and Iub load exceed one of at least corresponding separately load-threshold, then on existing bandwidth basis, reduce the VOIP bandwidth class step by step, also can skip a grade and reduce the VOIP bandwidth class according to the load-threshold of configuration, up to the VOIP bandwidth class is reduced to the assurance rate bandwidth, close cycle timer.
After the step 105, re-execute step 103.
But above-mentioned only is better embodiment of the present invention; be not to be used to limit protection scope of the present invention; any those skilled in the art of being familiar with will be appreciated that; all within the spirit and principles in the present invention scope; any modification of being done, equivalence replacement, improvement etc. all should be included within the scope of the present invention.
Claims (10)
1. the distribution method of a VOIP service bandwidth is characterized in that, comprising:
Steps A is according to the maximum rate of the VOIP business of RAB assignment with guarantee that speed is provided with the transmission channel bandwidth grade;
Step B, detected transmission channel loading situation;
Step C adjusts the grade of VOIP service bandwidth according to detected transmission channel loading condition, when described transmission channel load surpasses default thresholding, step by step or skip a grade and reduce the grade of described VOIP service bandwidth; When described transmission channel load is lower than default thresholding, then step by step or the grade of described VOIP service bandwidth of raising of skipping a grade.
2. the distribution method of a kind of VOIP service bandwidth as claimed in claim 1, it is characterized in that, also comprise before the steps A: core net assignment service quality QoS, carrying out RAB when setting up, according to the maximum rate of service quality QoS assignment VOIP business with guarantee speed.
3. the distribution method of a kind of VOIP service bandwidth as claimed in claim 2, it is characterized in that: in the steps A, the transmission channel bandwidth grade is set is specially the corresponding maximum rate of maximum bandwidth, guarantee the corresponding speed that guarantees of bandwidth, the intergrade bandwidth is selected successively according to the speed between maximum rate and the assurance speed.
4. as the distribution method of claim 1,2 or 3 described a kind of VOIP service bandwidths, it is characterized in that the load of transmission channel described in the step B is specially cell load, Iub load one of at least.
5. as the distribution method of claim 1,2 or 3 described a kind of VOIP service bandwidths, it is characterized in that, cycle timer is set, the opportunity of reduction VOIP service bandwidth grade is when exceeding default thresholding for and transmission channel load overtime when cycle timer.
6. as the distribution method of claim 1,2 or 3 described a kind of VOIP service bandwidths, it is characterized in that, cycle timer is set, the opportunity of rising VOIP service bandwidth grade is when being lower than default thresholding for and transmission channel load overtime when cycle timer.
7. the distribution method of a kind of VOIP service bandwidth as claimed in claim 4, it is characterized in that, also comprise between step B and the step C and judge whether cell load surpasses default load-threshold or whether the Iub load surpasses the default load-threshold of Iub, and then transmission channel loading condition described in the step C is this judged result.
8. the distribution method of a kind of VOIP service bandwidth as claimed in claim 7, it is characterized in that, when judging cell load or Iub load and exceed separately default load-threshold, step C is specially: step by step or skip a grade and reduce the grade of VOIP service bandwidth, and up to the VOIP service bandwidth is reduced to the assurance bandwidth.
9. the distribution method of a kind of VOIP service bandwidth as claimed in claim 4, it is characterized in that, also comprise between step B and the step C and judge whether cell load surpasses default load-threshold and whether the Iub load surpasses the default load-threshold of Iub, when judging cell load, Iub load and exceed one of at least default separately load-threshold, step C is specially: step by step or skip a grade and reduce VOIP service bandwidth grade, and up to the VOIP bandwidth is reduced to the assurance bandwidth.
10. the distribution method of a kind of VOIP service bandwidth as claimed in claim 4, it is characterized in that, also comprise between step B and the step C and judge whether cell load surpasses default load-threshold and whether the Iub load surpasses the default load-threshold of Iub, when judging cell load, Iub load and all be lower than default separately load-threshold, step C is specially: step by step or the rising VOIP service bandwidth grade of skipping a grade, and until being raised to maximum bandwidth.
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| CNB2006100616841A CN100484100C (en) | 2006-07-17 | 2006-07-17 | Distributing method for VOIP service band width |
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| CNB2006100616841A CN100484100C (en) | 2006-07-17 | 2006-07-17 | Distributing method for VOIP service band width |
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| CN100484100C true CN100484100C (en) | 2009-04-29 |
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| CN101420266B (en) * | 2007-10-24 | 2012-11-28 | 中兴通讯股份有限公司 | Pre-allocation resource location determining method based on semi-continuous scheduling |
| CN102149145A (en) * | 2010-02-10 | 2011-08-10 | 普天信息技术研究院有限公司 | Method for controlling non-scheduled resource in high-speed uplink packet access (HSUPA) service |
| CN102231697A (en) * | 2011-06-17 | 2011-11-02 | 瑞斯康达科技发展股份有限公司 | Bandwidth dispatching method of message queues as well as message reporting method and device |
| CN108271219B (en) * | 2016-12-30 | 2020-11-24 | 中国移动通信集团上海有限公司 | Method and device for controlling wireless network resources |
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Non-Patent Citations (2)
| Title |
|---|
| UMTS核心网中基于区分服务的QoS控制模型. 刘威，易波，毛珊.微电子学与计算机，第Vol.21卷第No.4期. 2004 |
| UMTS核心网中基于区分服务的QoS控制模型. 刘威，易波，毛珊.微电子学与计算机，第Vol.21卷第No.4期. 2004 * |
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