CN1756230A - Method for reducing real-time service time delay and time delay variation - Google Patents
Method for reducing real-time service time delay and time delay variation Download PDFInfo
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Abstract
The invention discloses a method for reducing the time delay and the time delay vibration of real-time service, which is characterized in that: said method comprises: setting the size of initial receiving buffer area; setting the data frame waiting time according to the sizes of initial receiving buffer area and the application service bandwidth; the sending end divides the network data package of real-time service into several wireless chain protocol data frames to be transmitted to the receiving end; the receiving end receives the wireless chain protocol data frames in the waiting time of data frame; regulating the size of buffer area according to the time delay vibration condition that the wireless chain protocol data frames reach the receiving end. The invention can reduce the time delay vibration caused by the chain circuit transmission, reduce the voice vibration of VoIP service and improve the voice quality of VoIP service.
Description
Technical field
The present invention relates to communication technical field, be specifically related to a kind of method that reduces real time business time delay and time delay shake.
Background technology
CDMA (code division multiple access) mobile radio communication is because its unit capacity is big, the high communication quality and be convenient to 3-G (Generation Three mobile communication system) upgrading the therefore positive important effect of play more and more in mobile communication system.Present cdma system, the same with GSM (global system for mobile communications) system, basically handling voice service, and along with professional flourish of Internet (internet), simple conversation can not have been satisfied user's requirement.A lot of users wish that mobile communication system can provide more function, and as entering Internet inquiry or transmission information, organize video conference, carry out remote diagnosis etc. by mobile phone when needed, this just requires cdma system that data service can be provided.
CDMA data service model wherein, has following network element as shown in Figure 1:
MSC: mobile switching centre; BSC: base station controller;
BTS: base transceiver station; VLR: VLR Visitor Location Register;
HLR: attaching position register; AC: AUC;
EIR: equipment identity register; SDH: SDH (Synchronous Digital Hierarchy);
PCF: packet control unit; SSP: Service Switching Point;
HA: home agent; FA: Foreign Agent;
AAA: mandate, authentication and charging center; MIP: mobile IP
PDSN: packet data serving node; GMSC: mobile gateway switching center;
ISDN: integrated services digital network; PLMN: Public Land Mobile Nerwork;
PSTN: public telephone network; INTERNET: internet;
Each main Network Element Function is as follows:
1, BSC (base station controller): finish the control and the management of base station; Call out establishment of connection and dismounting; Mobile management; By soft/direct-cut operation, for upper-layer service provides reliable and stable wireless connections; Power control; RRM.
2, PCF (Packet Control Function): finish R-P establishment of connection, maintenance and management.Because the anxiety of Radio Resource so just need to discharge radio channel resource when the user does not have transmission, reception on wireless channel when, keeps but PPP connects continuation.By handoff functionality, upper-layer service is shielded wireless mobility.
3, BTS (base station): mainly finish the transmission-receiving function of wireless signal, realize the communication between wireless system and the travelling carriage.
4, MSC (mobile switching centre): MSC is the core network device in CDMA2000 lattice network territory, and it finishes mobile networks' such as exchange the core business processing capacity of signaling process, the speech business of signaling process, the RRM of travelling carriage mobile management.During the Mobile Origination data service, MSC is responsible for setting up, safeguards, removes Traffic Channel.
5, MS (travelling carriage): be mobile subscriber equipment, can initiate by air interface, receipt of call, finish with the communication between the base transceiver station.When request msg was professional, MS initiated, safeguards and deletes PPP and is connected.
6, PDSN (packet service node): be the gateway device between packet data network and the CDMA2000 mobile communication system.PDSN is exactly a router in essence, but compare function with the router in PDN (packet data network) network enhancing is arranged again: strengthened the ability of in the CDMA2000 mobile communications network, transmitting packet, be that PDSN can be forwarded to the PDN output packet travelling carriage in the CDMA2000 network, the data service of the Mobile Originated access PDN in the CDMA2000 network also must realize by PDSN.When simple IP inserted, PDSN was that travelling carriage distributes dynamic IP addressing.When mobile IP inserted, PDSN served as the FA (external agent) of HA (home agent).
7, HA (home agent): be distinctive network element device in the mobile IP service network, actual is exactly a router, but has increased the location information function of preserving the current place of travelling carriage on the basis of router feature again.The major function of HA comprises three aspects:
(1) basic route forwarding function;
(2) create and safeguard data transmission tunnel between HA and the FA, set up the MBR (mobility binding record) between home IP address and the related IP address;
(3) function of the current latest position information of preservation travelling carriage is similar to the HLR (attaching position register) in the mobile communication system.
8, AAA (Authorization, Authentication and Accounting, aaa server): the function of the aaa server in the CDMA2000 packet domain core network network and the function of the aaa server in the fixed data network are on all four, are mainly used in mobile data user's mandate, authentication and charging.
Interface between aaa server and the PDSN is the IP protocol interface of standard, and the consensus standard of following is RFC2138 and RFC2139.This interface is the standard interface between the fixed data network.
9, IWF (IWF module): be used for realizing that agreement and data are adaptive.IWF equipment can be used to connect two networks (as network adapter) or a terminal and a network (as terminal adapter).IWF provides following mechanism: signaling is adaptive, media is controlled, media is adaptive.
Mobile subscriber's the data service based on Circuit Service is sent by base transceiver station, by base station controller business is sent to mobile switching centre then, mobile switching centre selects to transmit by Public Land Mobile Nerwork according to the characteristic of this data service again, or selects public telephone network or integrated services digital network to transmit by mobile gateway switching center.
Mobile subscriber's the data service based on Packet Service is sent by base transceiver station, by packet control unit business is sent to packet data serving node then, by mobile IP this data service is entered the Internet and transmits.
Because data streaming service has certain error rate requirement under wireless communications environment, SDU (Service Data Unit) carries out RLP agreement and retransmission protocol, promptly when the recipient requires the retransmission of lost Frame, sends and retransmits this Frame that exists in the buffer just now.RLP carries out buffering can bring 1 to 2 second stand-by period with repeating transmission, if RLP adopts frame length 20ms as retransmitting elementary cell, a simple eight bit byte stream interface is arranged between the receiving-transmitting sides.
At present, VoIP (Voice over Internet Protocol) occupies increasing proportion in the Internet types of applications, and VoIP is a kind of based on IP phone, and releases the technology of corresponding value-added service.It is a transmission platform with the IP packet switching network, a series of special processings such as the voice signal to simulation compresses, packing, make it to adopt connectionless UDP (user datagram protocol) agreement to be transmitted, its maximum advantage is to adopt the environment of Internet and global ip interconnection widely, and, better service more than traditional business are provided.The business such as transmission voice, fax, video and data that VoIP is can be on IP network cheap are transmitted and the storage forwarding of various information etc. as unified message, virtual telephony, virtual speech/fax mailbox, directory enquiry service, Internet call center, Internet call manager, video conference, ecommerce, fax storage.The extensive use of VoIP, the problem that its service quality guarantees is also paid close attention to by industry day by day.VoIP is typical real-time service, for QoS (service quality) very high requirement is arranged.The user is when using the VoIP business, and what be concerned about most is exactly the voice quality problem of conversation.But present VoIP lacks QoS to be guaranteed.
The realization of VoIP is based on IP network, and IP network can be the local area network (LAN) or the Internet of enterprises.In general, voice messaging is sent by PBX (PBX) from transmitting terminal, and gateway, local area network (LAN), router by the source end enter IP-based network again, arrives the PBX of destination after the outflow IP network, arrives recipient's phone at last.In this process, there are several places can cause the time delay of speech data.At first, analog voice signal is sent to voice gateways by PBX, and gateway can add certain time delay when giving speech signal coding, and concrete time delay size is relevant with the speech coder that gateway adopts.To be packaged into packet after speech signal coding is intact and be sent to router through LAN (local area network (LAN)).Router can be according to the different time delays that produce in various degree of packet size and network insertion circuit speed.After packet enters IP network, also will could arrive the outlet of network through one or more router routes, these routers are also increasing time delay in varying degrees.The size of this time delay is uncertain, and it is mainly determined to number, the processing speed of each router and the load of each router of the router that flows out network institute road warp from entering network by packet.Time delay when total time delay has comprised time delay that the voice transfer packet produces and the IP network by wide area by local area network (LAN) the time.
Circuit Switching Network to the influence of this class quality of real-time service of VoIP mainly from two aspects: the processing delay of Circuit Switching Network and Circuit Switching Network cooperate with signaling between the gateway.Usually, provide QoS (service quality) to guarantee that mechanism reduces the influence to the VoIP QoS by Circuit Switching Network self.
In the existing network system,, make the time delay of VoIP to be guaranteed because the QoS of CDMA system guarantees that mechanism is not perfect at present based on CDMA.In addition, be used on the air interface grouped data and use RLP (radio link protocol) agreement to transmit user data reliably between MS and SDU (Service Data Unit), the error rate of assurance grouped data on wireless network can approach fixed network.When transmitting the VoIP business, when data receiving terminal " found " wrong after, the recipient can require transmit leg to carry out transfer of data again, if necessary, carry out repeatedly, till the reception data are entirely true, the recipient need wait for that packet collection Qi Houcai gives the last layer encoder, therefore the mechanism that should guarantee has increased the time delay and the delay variation of VoIP business for the VoIP business, has influenced such quality of service.
Summary of the invention
The purpose of this invention is to provide a kind of method that reduces real time business time delay and time delay shake, to solve the problem that can't guarantee the real time business time delay in the prior art, reduce owing to time delay and delay variation based on the real time business that causes in circuit and the packet data transmission process.
For this reason, the invention provides following technical scheme:
A kind of method that reduces real time business time delay and time delay shake, described method comprises:
A, the initial size that receives buffering area of setting;
B, according to described initial reception buffering area and the application service bandwidth the big or small setting data frame stand-by period;
C, transmitting terminal are split as a plurality of radio link protocol Frames with the network layer data bag of described real time business and send to receiving terminal;
D, described receiving terminal receive described radio link protocol Frame at described Frame in the stand-by period;
E, the delay variation situation that arrives at described receiving terminal according to described radio link protocol Frame are dynamically adjusted the size of described reception buffering area.
Described steps A is specially:
Set the initial size that receives buffering area according to packet loss, time delay and delay variation situation that type of service allows.
Described step D comprises:
First radio link protocol Frame of D1, the described network layer data bag of reception;
D2, will wrap in the whole network layer data that described Frame received in the stand-by period and pass to described receiving terminal network layer.
Described step D2 comprises:
D21, setting timer, the timing of described timer is the described Frame stand-by period;
D22, receive and start described timer behind described first radio link protocol Frame;
D23, behind described timer expiry, judge whether to receive whole radio link protocol Frames of described network layer data bag;
If D24 all receives, then the network layer data that receives is wrapped and passed to described receiving terminal network layer;
If D25 does not all receive, then this network layer data bag is abandoned.
Described step e comprises:
Set increase threshold value and minimizing threshold value that described radio link protocol Frame arrives at the delay variation statistical value of described receiving terminal;
When the delay variation statistical value that arrives at described receiving terminal when described radio link protocol Frame surpasses described increase threshold value, increase described reception buffering area.
When the delay variation statistical value that arrives at described receiving terminal when described radio link protocol Frame is lower than described minimizing threshold value, reduce described reception buffering area.
Described network layer data bag comprises:
Point-to-point protocol packet, Internet Protocol packet, User Datagram Protoco (UDP) packet, RTP packet.
Described transmitting terminal comprises portable terminal, base station, and its corresponding receiving terminal is respectively base station, portable terminal.
When described transmitting terminal was described portable terminal, described step C also comprised:
With described real time business data encapsulation is the network layer data bag;
Described network layer data bag is transmitted to described wireless link protocol layer.
By above technical scheme provided by the invention as can be seen, the present invention is when transmitting real time business, by the stand-by period of restriction RLP (radio link protocol) layer receiving data frames, avoided effectively because the admission control of certain IP packet causes follow-up IP packet can not in time be uploaded to the IP layer, and because admission control is introduced long time delay, thereby reduced the IP packet delay variation in the transmission course aloft.In addition, the present invention also by dynamically adjust the mode that RLP receives buffering area according to the jitter conditions of Frame arrival time, has reduced the voice jitter of VoIP business effectively, thereby has improved the voice quality of VoIP business.
Description of drawings
Fig. 1 is a CDMA circuit type data service model;
Fig. 2 is the flow chart of the inventive method;
Fig. 3 is that the IP packet splits the reception schematic diagram at the RLP layer;
Fig. 4 utilizes timer to realize the flow chart of restricting data frame stand-by period in the inventive method;
Fig. 5 is a protocol stack structure schematic diagram of realizing VoIP on cdma network;
Fig. 6 is based on the voip call processing procedure of Circuit Service in the cdma system;
Fig. 7 is based on the voip call processing procedure of Packet Service in the cdma system.
Embodiment
Core of the present invention is that portable terminal and base station system are in data transmission procedure, stand-by period when restriction RLP (radio link protocol) layer (end side or base station side) receives the network layer data bag, the complete data packet of collecting in the given time is uploaded to the receiving terminal network layer, if do not receive the Frame of packet in the given time, just with this data packet discarding, to avoid owing to admission control is introduced long time delay; Simultaneously, dynamically adjust RLP according to the jitter conditions of Frame arrival time and receive buffering area, to guarantee the voice quality of VoIP business.
The present technique field personnel know, WCDMA (Wideband Code Division Multiple Access (WCDMA))/CDMA2000 is a 3-G (Generation Three mobile communication system), adopted and the similar structure of second generation mobile communication system, on system configuration and function, comprised Radio Access Network (RAN) and core network (CN).Wherein, Radio Access Network is used to handle all and the wireless function associated that comprises the management of Radio Resource; Core network is responsible for the interior all voice calls of treatment system and is connected with data, and the exchange of realization and external network and route.
Core network is from being divided into two subsystems in logic: circuit commutative field (CS) and packet-switched domain (PS).Traditional circuit-switched service is handled in the CS territory, and each communication need take some resources and set up a special-purpose link, as speech business; Packet switching service is handled based on packet switching network in the PS territory, and signaling and data all are to transmit in Packet Based Network with data packet form, do not need to set up dedicated link, and each grouping all oneself is looked for route.
Different Business Nature differences, the quality of service QoS (service quality) of requirement is also different.Quality of service QoS can comprise following index: the error rate, delay, throughput, reliability and fail safe etc.For data service,, can be divided into: session service, stream class service, interactive service and backstage business according to the feature of QoS.Wherein, session and stream class service carrying real time business, mutual and backstage service bearer non-real-time service.
For voice service, we know that the factor that influences speech quality mainly contains: coding method, time delay, delay variation, packet loss, bit error, protocol processes.In IP-based cdma system, can speech quality be described with parameters such as service availability, delay, variable delay, throughput, packet loss.
Therefore from user's demand and system design goal, the quality of service requirement of speech packet there is the following aspects:
(1) to the quality assurance of speech packet time delay.
It is 300ms that the call voice transmission comes and goes maximum delay, and the 150ms One Way Delay is an acceptable in other words.
(2) reduce thrashing and time delay.
With the base station portion is example, following situation can occur: the Frame that sends from same MS (portable terminal) is uniform, because the inequality of IP network traffic carrying capacity, after the different sectors or different B TS arrival BSC (base station controller) via BTS (base station system), the time interval that they arrive BSC then no longer is even interval, and may frame losing, it seems from this side of BSC so thrashing just to occur.This problem can have influence on the process of normal talking, solves badly, can cause the interrupted or distortion of speech.
(3) improve speech quality.
For VoIP class business,, and not high especially for the sensitivity of packet loss because its sensitivity to time delay and delay variation is higher.The present technique field personnel know, RLP (radio link protocol) agreement is a kind of packet retransmissions present agreement, and promptly when the recipient required the retransmission of lost Frame, emission side retransmitted this Frame that exists in the buffer.Execution RLP agreement makes and transmit user data reliably between MT and base station system, guarantee that the error rate of grouped data on wireless network can approach fixed network, but the mechanism that should guarantee is for the VoIP business, be the mechanism that a kind of time delay and delay variation increase, RLP carries out buffering can bring 1 to 2 second stand-by period with repeating transmission.Therefore, at this class business, on cdma network, restriction waits for that the time of RLP re-transmission will reduce the time delay and the time delay shake of VoIP business effectively.
The present invention is exactly when the MS of cdma system and BSC (base station controller) receive the RLP Frame, reduces the real time business time delay by the restricting data frame stand-by period.
In order to make those skilled in the art person understand the present invention program better, the present invention is described in further detail below in conjunction with drawings and embodiments.
With reference to Fig. 2, Fig. 2 is the flow chart of the inventive method, may further comprise the steps:
Step 201: set the initial size that receives buffering area according to packet loss, time delay and delay variation situation that type of service allows.
Step 202: according to the big or small setting data frame stand-by period of initial reception buffering area and application service bandwidth.
Step 203: transmitting terminal obtains the network layer data bag of real time business.Described transmitting terminal comprises: portable terminal and base station controller, certainly, its corresponding receiving terminal is respectively base station controller and portable terminal.Described portable terminal can be the calling subscriber, also can be the called subscriber.
For the VoIP business,
Caller MS (portable terminal) is to the data transmission procedure of MSC (base station controller), at first by speech coding the voice of VoIP business are packaged into the network layer data bag by the vocoder of MS, such as, be the PPP information bag by PPP (point-to-point protocol) protocol encapsulation; By IP (Internet Protocol) protocol encapsulation is the IP packet; By UDP (User Datagram Protoco (UDP)) protocol encapsulation is the UDP message bag; Also can be encapsulated as the RTP packet by RTP (RTP) layer protocol.Then, this network layer data bag is transmitted to the RLP layer of MS.
In the data transmission procedure of MSC to MS, at first directly obtain the network layer data bag of called MS by BSC.Equally, according to the difference of tunneling, can be PPP information bag, IP packet, UDP message bag and RTP packet.Then, this network layer data bag is transmitted to the RLP layer of BSC.
Step 204: the network layer data bag is split as a plurality of radio link protocol Frames by the RLP layer.
As shown in Figure 3: an IP packet is split as 6 RLP data frame transfer at the RLP layer, the RLP layer of BSC side (or end side) sends this 6 Frames in order, and the RLP layer of terminal (or BSC side) collects just this IP to be wrapped behind 6 Frames of IP bag passes to terminal (or BSC) ip protocol layer.
Usually,, make the transmission time shake big, and the RLP layer only collect 6 Frames and just IP can be wrapped and pass to the IP layer because wireless transmission may cause the out of order of RLP layer data frame or loses.Can cause IP packet 1 can not be uploaded to the IP layer as the A admission control, and can cause IP packet 2 also can not be uploaded to the IP layer.The RLP layer of terminal can be waited for the arrival of A Frame always, and the A Frame just can be uploaded to the IP layer with IP packet 1 and IP packet 2 after arriving according to the order of sequence.The wait of RLP layer data frame can directly cause the shake of the propagation delay time and the time delay of IP packet, influences the voice quality of VoIP business.
For the propagation delay time that reduces the IP packet and the shake of time delay, the inventive method in step 302 according to the initial big or small setting data frame stand-by period that receives buffering area and application service bandwidth.
In the setting data frame stand-by period, receive the RLP Frame, handle by following mode:
Refer again to Fig. 2,
Enter step 205: first RLP Frame that receives the network layer data bag.
Then, enter step 206: receive other follow-up RLP Frames of this network layer data bag in the stand-by period at Frame.
Step 207: will wrap in the whole network layer data that Frame received in the stand-by period and pass to the receiving terminal network layer.
Step 208: will not receive whole network layer data bag in the stand-by period at Frame and abandon.
Step 209: the delay variation situation that arrives at receiving terminal according to the RLP Frame is dynamically adjusted the size that receives buffering area.
In the process of above-mentioned reception RLP Frame, can realize by timer the restriction of Frame stand-by period, concrete with reference to flow process shown in Figure 4:
At first, in step 401: set timer, the Frame stand-by period of the timing of described timer for setting;
Step 402: start timer after receiving first radio link protocol Frame.
Step 403: behind the timer expiry, judge whether to receive whole radio link protocol Frames of network layer data bag.
If all receive, then enter step 404: the network layer data that receives is wrapped passed to the receiving terminal network layer.
Otherwise, enter step 405: this network layer data bag is abandoned.
In the process of above-mentioned reception RLP Frame, the radio link protocol Frame that receives need be stored in the initial reception buffering area.The packet loss that the size of buffering area allows according to type of service, time delay and delay variation situation are set.Behind first RLP Frame that receives the network layer data bag, this Frame is stored in the reception buffering area, order receives follow-up RLP Frame then, and the follow-up RLP Frame that will receive successively stores in this buffering area.Receive whole follow-up RLP Frames in stand-by period at the Frame of setting, a whole network layer data that then will be stored in this buffering area is wrapped the network layer of passing to receiving terminal.Because in receiving course, may receive wrong RLP Frame, such as, when wireless signal was very weak, the delay variation of network layer data bag can be very big.For equilibrium receives time delay and the delay variation that produces in the RLP Frame process, guarantee quality of service better, the present invention also dynamically adjusts the size that receives buffering area according to the delay variation situation that the RLP Frame arrives at receiving terminal.
The mode of dynamically adjusting the size that receives buffering area is as follows:
Setting wireless link protocol Frame arrives at the delay variation threshold value of receiving terminal;
When the delay variation that arrives at receiving terminal when the radio link protocol Frame surpasses this threshold value, suitably increase and receive buffering area;
The delay variation that arrives at receiving terminal when the radio link protocol Frame recovers the initial size of this reception buffering area during less than this threshold value certain limit.
Such as, when wireless signal is very weak, can cause the delay variation of IP packet very big, at this moment, mobile phone can suitably strengthen the reception buffering area of RLP Frame; When mobile phone roaming during to the stronger receiving area of signal, the delay variation of IP packet diminishes, and at this moment, the reception buffering area that mobile phone can reduce the RLP Frame is the size of initial setting.
Fig. 5 is a protocol stack schematic diagram of realizing VoIP on cdma network:
Data transmission procedure is as follows:
(1) data transmission procedure of caller MS to BSC
At first, the vocoder of terminal is packaged into PPP/IP/UDP/RTP packet (network layer data bag) by speech coding, is sent to network layer (PPP/IP/UDP/RTP layer), and network layer is transmitted to the RLP layer with packet.The RLP layer is split as a plurality of RLP Frames with packet, is sent to the RLP radio link layer of BSC side then in order.
The process that BSC receives the RLP Frame is as follows:
At first, BSC receives first RLP Frame of network layer data bag and stores into and receives in the buffering area.Can preestablish the adjustment strategy of this buffering area, promptly dynamically adjust the size that RLP receives buffering area according to the jitter conditions of Frame arrival time.Start timer then, order receives the follow-up RLP Frame of this network layer data bag.Behind timer expiry, if also have Frame not receive in this network layer data bag, abandon this packet so, and the follow-up data that will receive is wrapped and is passed to network layer.Avoided because the admission control of certain network layer data bag causes the follow-up data bag can not in time be uploaded to network layer.Avoided effectively because admission control is introduced excessive time delay, defined network layer data and wrap in delay variation in the aerial transmission course.
(2) BSC is to the data transmission procedure of called MS side
At first, BSC receives the RLP Frame of caller MS, and the complete network layer data bag that receives is sent to called MS.
The processing procedure of the RLP protocol layer receiving data frames of called MS is as follows:
At first, called MS receives first RLP Frame of network layer data bag and stores into and receives in the buffering area.Equally, also can preestablish the adjustment strategy of this buffering area, promptly dynamically adjust the size that RLP receives buffering area according to the jitter conditions of Frame arrival time.Start timer then, order receives the follow-up RLP Frame of this network layer data bag.When timer expiry, if also have Frame not receive in this network layer data bag, abandon this packet so, and the follow-up data that will receive is wrapped and is passed to network layer.After network layer receives this packet, this packet is carried out decapsulation, obtain VoIP speech business data, by vocoder these data are decoded then.
For example, Fig. 6 be in the cdma system based on the call flows of the VoIP of Circuit Service, comprise following process:
1, sets up Traffic Channel between MS (portable terminal) and BS (base station);
2, it is synchronous to carry out RLP;
3, MS and IWF (between the interworking module) carry out the PPP/LCP/IPCP/ negotiation;
4, setting up TCP connects;
5, Application interface (application interface) parameter transmission;
6, it is synchronous to carry out Modem between IWF and the PSTN (public switched telephone network);
7, carry out end-to-end ppp negotiation between MS and the PSTN;
8, setting up end-to-end TCP connects;
9, setting up application layer connects.
At first, set the initial size that receives buffering area according to packet loss, time delay and delay variation situation that type of service allows, and, in the setting data frame stand-by period, receive the RLP Frame according to big or small setting data frame stand-by period of initial reception buffering area and application service bandwidth.
Make a call by MS (portable terminal), MS at first sends call request information (comprising business option) to BS (base station), undertaken distributing wireless channel behind customer location and the identity validation by BSC/MSC (base station controller/mobile switch), and send confirmation to MS.Initialization RLP (radio link protocol) then, LCP (LCP) layer carries out link initialization and makes each protocol synchronization, and IPCP (IP control protocol) arranges the IP address.After TCP/IP (transmission control protocol/Internet Protocol) is initialised, setting up TCP connects, transmit Application interface parameter, BSC or MSC give MS distributing IP address, then Modem (modulator-demodulator) instruction that keeps in advance in the MS adapter is transferred to BSC or MSC by protocol package, carry out end-to-end ppp negotiation; Set up end-to-end TCP then and connect, set up application layer at last and connect.So far, then can carry out corresponding data, services.
When carrying out the real time business transmission, split transmission according in the invention described above network layer data being wrapped in RLP layer fractionation mode, receive according to reception buffering area of setting and Frame reception stand-by period during reception.Reduce real time business time delay and time delay shake with this.And, for equilibrium receives time delay and the delay variation that produces in the RLP Frame process, guarantee quality of service better, need dynamically adjust the size that receives buffering area according to the delay variation situation that the RLP Frame arrives at receiving terminal.
The present invention is not only applicable to the real time business on the circuit type data service, is equally applicable to the real time business on the Packet data service.
In the real-time service transmission based on Packet data service, time delay and time delay shake end to end is the principal element that influences VoIP class quality of real-time service.Time delay is end to end only considered the time delay that the bag of voice-bearer information is experienced usually from the emission system to the receiving system.According to different offered load situations, postponing a meeting or conference in the time of end to end changes.In VoIP, delay variation generally is meant the difference of the end-to-end time delay of two continuous voice packets in the voice flow.Delay variation has remarkable influence to the performance of the application such as VoIP (other also comprise video playback etc.) that need the regularization transmission package.Specifically, its work that voice packet is rebuild according to original series and cyclic pattern has negative effect.This moment, the maximum delay shake was an important indicator weighing performance.Because IP bag itself just exists delay variation, want that the delay variation of elimination voice packet in network is impossible.Therefore need manage to alleviate the adverse effect that delay variation is used VoIP.
Adopt the invention described above method, the packet loss, time delay and the setting of delay variation situation that allow according to type of service initially receive the size of buffering area and the big or small setting data frame stand-by period that basis initially receives buffering area and application service bandwidth.
Like this, after receiving voice packet, do not play immediately, but temporarily be retained in the initial reception buffering area, arrive, again the bag that accumulates in the buffering area is carried out rule and play, thereby delay variation is reduced to minimum up to predetermined reproduction time.
If set bigger reception buffering area, then can reduce packet loss, but can make the bag that early arrives introduce long time delay; On the contrary, if set less reception buffering area,, can influence packet loss though reduced the influence of transfer delay and time delay shake.
Therefore, for balanced caching delay and packet loss, the present invention is according to the big or small setting data frame stand-by period of initial reception buffering area and application service bandwidth.If in waiting time, collect the total data frame of a bag, then this bag is passed to the IP layer, otherwise abandon this bag.Like this, by limiting the stand-by period of Frame, guaranteeing that packet loss satisfies the influence that reduces delay variation under the situation of certain index effectively.
Owing to be engraved in variation during wireless environment, like this, give time delay that the VoIP business brings and delay variation also in constantly changing, for guaranteeing that terminal is under various Radio Resource situations, can both obtain better tone quality, the present invention arrives at the time delay of receiving terminal and the size of the reception buffering area that the delay variation situation is dynamically adjusted initial setting according to Frame, to guarantee the compromise of packet loss and time delay.Such as, when the wireless environment variation, the delay variation of IP packet can be very big, at this moment, can suitably strengthen the reception buffering area, exchanges the minimizing of packet loss for time delay and delay variation; Otherwise when wireless environment was fine, the delay variation of IP packet diminished, and at this moment, can suitably reduce to receive buffering area, exchanged the minimizing of time delay and delay variation for certain packet loss.Like this, by to receiving the dynamic adjustments of buffering area, the time delay of VoIP business and delay variation and packet loss all are controlled in certain scope, can guarantee the quality of VoIP business better.
For example, Fig. 7 shows in the cdma system call flows based on the VoIP of Packet Service, comprises following process:
A.MS sends the originating information of belt 2 authorization request aloft on the access channel of interface to BS, with the request Packet data service.
B.BS confirms that by send the base station to MS instruction confirms the reception to originating information.
C.BS constructs business request information, puts it in complete layer 3 message, mails to MSC, simultaneously opening timing device T303.
D.MSC sends assignment request message to BS, with the request distributing radio resource, and while opening timing device T10.For Packet data service, do not need to distribute the land circuit between MSC and BS.BSC timeing closing device T303.
E.BS and MS initiate the creation facilities program (CFP) of radio traffic channel.
F. connect in order to set up A8, BS sends A9-to PCF and builds-A8 message and opening timing device TA8-foundation.
G. this step is carried out the A10/A11 establishment of connection.
H. after receiving A9-foundation-A8 message, PCF sets up the A8 connection and sends A9-connection-A8 message.
I. after receiving A9-connection-A8 message, BS stops timer TA8-foundation and sends assignment finishing message.MSC timeing closing device T10.This step can take place in any time after Radio Link is set up.
J. between MS and PDSN, carry out the registration of PPP establishment of connection program and mobile IP.
K. transmit Packet Service.
When carrying out transmitting based on the real time business of Packet Service, split transmission according in the invention described above network layer data being wrapped in RLP layer fractionation mode, receive the stand-by period according to the reception buffering area of setting and Frame during reception and receive.Reduce real time business time delay and time delay shake with this.And, for equilibrium receives time delay and the delay variation that produces in the RLP Frame process, guarantee quality of service better, need dynamically adjust the size that receives buffering area according to the delay variation situation that the RLP Frame arrives at receiving terminal.
Though described the present invention by embodiment, those of ordinary skills know, the present invention has many distortion and variation and do not break away from spirit of the present invention, wish that appended claim comprises these distortion and variation and do not break away from spirit of the present invention.
Claims (8)
1, a kind of method that reduces real time business time delay and time delay shake is characterized in that described method comprises:
A, the initial size that receives buffering area of setting;
B, according to described initial reception buffering area and in please service bandwidth the big or small setting data frame stand-by period;
C, transmitting terminal are split as a plurality of radio link protocol Frames with the network layer data bag of described real time business and send to receiving terminal;
D, described receiving terminal receive described radio link protocol Frame at described Frame in the stand-by period;
E, the delay variation situation that arrives at described receiving terminal according to described radio link protocol Frame are dynamically adjusted the size of described reception buffering area.
2, method according to claim 1 is characterized in that, described steps A is specially:
Set the initial size that receives buffering area according to packet loss, time delay and delay variation situation that type of service allows.
3, method according to claim 1 is characterized in that, described step D comprises:
First radio link protocol Frame of D1, the described network layer data bag of reception;
D2, will wrap in the whole network layer data that described Frame received in the stand-by period and pass to described receiving terminal network layer.
4, method according to claim 3 is characterized in that, described step D2 comprises:
D21, setting timer, the timing of described timer is the described Frame stand-by period;
D22, receive and start described timer behind described first radio link protocol Frame;
D23, behind described timer expiry, judge whether to receive whole radio link protocol Frames of described network layer data bag;
If D24 all receives, then the network layer data that receives is wrapped and passed to described receiving terminal network layer;
If D25 does not all receive, then this network layer data bag is abandoned.
5, according to each described method of claim 1 to 4, it is characterized in that described step e comprises:
Set increase threshold value and minimizing threshold value that described radio link protocol Frame arrives at the delay variation statistical value of described receiving terminal;
When the delay variation statistical value that arrives at described receiving terminal when described radio link protocol Frame surpasses described increase threshold value, increase described reception buffering area.
When the delay variation statistical value that arrives at described receiving terminal when described radio link protocol Frame is lower than described minimizing threshold value, reduce described reception buffering area.
6, method according to claim 1 is characterized in that, described network layer data bag comprises:
Point-to-point protocol packet, Internet Protocol packet, User Datagram Protoco (UDP) packet, RTP packet.
7, method according to claim 1 is characterized in that, described transmitting terminal comprises portable terminal, base station, and its corresponding receiving terminal is respectively base station, portable terminal.
8, method according to claim 7 is characterized in that, when described transmitting terminal was described portable terminal, described step C also comprised:
With described real time business data encapsulation is the network layer data bag;
Described network layer data bag is transmitted to described wireless link protocol layer.
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