CN101030924A - Method for adapting dynamic bandwidth - Google Patents

Method for adapting dynamic bandwidth Download PDF

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Publication number
CN101030924A
CN101030924A CNA2006100578905A CN200610057890A CN101030924A CN 101030924 A CN101030924 A CN 101030924A CN A2006100578905 A CNA2006100578905 A CN A2006100578905A CN 200610057890 A CN200610057890 A CN 200610057890A CN 101030924 A CN101030924 A CN 101030924A
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bandwidth
bag
rtcp
stream
medium
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CN100589436C (en
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李凤军
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Gu Haiyan
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ZTE Corp
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Abstract

The method is used in a stream media system with the media content having multi different coding bandwidths. After the terminal establishes the stream protocol session with the stream media system, when the stream media system receives the real-time control protocol (RTCP), the method comprises the following steps: estimating the transmission bandwidth and records it; taking the average value of several previous transmission bandwidths as the target bandwidth; deciding if the current packet-losing rate reported by RTCP is over a preset threshold; if yes, according to the target bandwidth, selecting a media content having coding bandwidth being the closest to t he target bandwidth.

Description

The method that a kind of dynamic bandwidth is adaptive
Technical field
The present invention relates to mobile communication streaming media service field, relate in particular to the adaptive method of a kind of dynamic bandwidth.
Background technology
Streaming media service is the application of a kind of employing " stream " mode transmitting media content, along with the construction of broadband network, is used on a large scale gradually.The outstanding feature of streaming media service is a real-time, sets up the initial stage at flow media session, carries out of short duration media content buffering, and the main cause of buffering is to be used to eliminate the influence to the media play effect that network fluctuation causes.Under subsequent network situation in order, interrupting can not appear in streaming media playing, finishes up to playing.But under the actual network condition, the network bandwidth changes very violent, reduction that might be at double, and the media content buffering of short time can not be eliminated the influence of network bandwidth fluctuation to the media play effect.If increase the initial buffer time, the user need cause the user to produce boredom through the wait of long period.
Compare streaming media service, downloading service is that downloading service at first downloads to playback terminal fully with media content, just plays then with the example of initial buffer time maximization.The advantage of streaming media service do not need to be that medium are downloaded to this locality fully and can to play, and improves the promptness that the user uses.But also strengthened the dependence of streaming media service to the network bandwidth.If the network bandwidth can not satisfy the encoded bandwidth requirement, the phenomenon that media play postpones then can appear, even the packet loss phenomenon.Situations such as packet loss occur, then can have influence on the content of broadcast, as anamorphose, mosaic etc.
Usually the basic network according to streaming media service can be divided into the streaming media service of cable network and the streaming media service of wireless network.Because the increase of network traffics, the network bandwidth becomes competitive resource.No matter all there are the adaptive problem of the network bandwidth in cable network or wireless network to streaming media service, relatively cable network, because the environment of wireless network is more abominable, the network bandwidth changes more violent, therefore, more needs the application of network bandwidth adaptation technique.The network bandwidth is just adaptive, according to the actual conditions of network, selects the media content of suitable encoded bandwidth to send to playback terminal.The real-time that keeps media content to send.Streaming media service is divided into program request and live two kinds of basic services, and program request is meant the business of mobile phone terminal by the media file of stream protocol broadcast appointment, and the live mobile phone terminal that is meant is play the business of real-time media stream by stream protocol.Both do not have difference on media delivery, and just the live broadcast service user can't independently select playing progress rate.
On the adaptive problem of the network bandwidth, a lot of research is arranged, some relevant patent files are also arranged.From the patent situation, roughly have dual mode to realize that the network bandwidth is adaptive: method one, the mutual agreed bandwidth of streaming server and playback terminal selects a kind of coded media content of coupling to send to playback terminal by server; Method two, the RTCP that streaming server returns according to playback terminal (RTCP Real-time Transport Control Protocol) bag message judges that the network bandwidth changes, and selects suitable coded media content to send.
The realization of method one needs to adopt between streaming server and the playback terminal identical agreement protocol, can adopt identical agreement protocol between the streaming server of same producer and the playback terminal, consult but then can't carry out such bandwidth between the streaming server of different manufacturers and the playback terminal.Therefore, the practicality of method one and versatility are relatively poor.
Method two is realized according to the standard that existing most of playback terminals all return the RTCP bag, the simple RTCP that returns that relies on carries out bandwidth adaptation, just simple loss situation from the RTCP bag, RTP (RTP) the packet loss situation of report in the RTCP bag, and the transmission delay situation, judge whether to carry out bandwidth adaptation.The amplitude that encoded bandwidth (following abbreviation medium bandwidth) that can not clear and definite media content is adjusted will inevitably cause because bandwidth is adjusted improper and can not be guaranteed the effect of user's playing stream media.And reach which kind of degree for RTP packet loss degree and transmission delay degree and just begin to carry out the adjustment of medium bandwidth and do not have clear and definite saying, can only carry out according to so-called empirical value, and network condition change various, the effect that empirical value is difficult to play.Therefore, the exploitativeness of second method is poor, and it is low to adjust efficient, and is difficult to carry out the adjustment of medium bandwidth in advance, avoids user's result of broadcast to be affected.
Summary of the invention
The technical problem to be solved in the present invention provides the adaptive method of a kind of dynamic bandwidth, can better adjust the medium bandwidth, guarantees the effect of user's playing stream media.
In order to solve the problems of the technologies described above, the invention provides the adaptive method of a kind of dynamic bandwidth, be applied to have the stream media system of the media content of multiple different coding bandwidth, after terminal and stream media system are set up the stream protocol session, stream media system is handled when receiving RTCP Real-time Transport Control Protocol RTCP bag according to the following steps:
(a) estimate transmission bandwidth and record, again the transmission bandwidth of several times estimation is before averaged as the target bandwidth;
(b) judge whether that described RTCP bag report present packet loss goes beyond the limit, if select an encoded bandwidth media content immediate to send with it according to described target bandwidth.
Further, said method also can have following characteristics: in the described step (a), after terminal and stream media system are set up the stream protocol session, preserve the medium amount of bandwidth of choosing for the first time, and be high bandwidth with this bandwidth, in the described step (b), RTCP bag present packet loss does not go beyond the limit as described, judge again current transmission bandwidth and target bandwidth whether basically identical and medium bandwidth also can increase, in this way, select the media content of upper level encoded bandwidth to send, finish this time to handle.
Further, said method also can have following characteristics: in the described step (b), RTCP bag present packet loss does not go beyond the limit as described, judge whether basically identical of current transmission bandwidth and target bandwidth again, if and the medium bandwidth can also increase, select the media content of upper level encoded bandwidth to send; Otherwise the medium bandwidth of described target bandwidth and current transmission differs bigger, reselects medium bandwidth to be sent, and then the medium bandwidth of judging described target bandwidth and current transmission basically identical whether.
Further, said method also can have following characteristics: in the described step (a), stream media system only after receiving legal RTCP bag, is just estimated transmission bandwidth and record.
Further, said method also can have following characteristics: in the described step (a), if stream media system is received the RTCP bag in RTCP bag maximum duration interval, comprise the RR bag in the described RTCP bag, and the part that the corresponding RTP of described RTCP bag wraps corresponding to the medium that transmitting surpasses the setting thresholding, and then described RTCP bag is legal RTCP bag.
Further, said method also can have following characteristics: in the described step (a), stream media system is when receiving the RTCP bag, be set the longest blanking time that receives next RTCP bag, if next RTCP wraps in to return in the longest described blanking time and comprise RR and wraps, then estimate transmission bandwidth and record; Otherwise, judge whether to have the media content that hangs down the level encoder bandwidth, if having, select the media content of low level encoder bandwidth to send, processing this time finishes; If the media content of low level encoder bandwidth does not stop whole media content and sends.
Further, said method also can have following characteristics: in the described step (a), if next RTCP wraps in to return in the longest described blanking time and comprise RR and wraps, to judge also whether the corresponding RTP bag of described RTCP bag surpasses the setting thresholding corresponding to the part of the medium that transmitting, if estimate transmission bandwidth and record again; Otherwise directly finish.
Further, said method also can have following characteristics: in the described step (a), stream media system according to the size of average each the RTP bag that calculates and the RTP bag number that streaming media playing terminal actual reception arrives, obtains the byte number that the streaming media playing terminal receives when receiving the RTCP bag; The byte number that receives according to the time interval and the stream media terminal of RTCP bag again, estimation transmission bandwidth and record.
Further, said method also can have following characteristics: in the described step (a), stream service system is when receiving a RTCP bag, the RTP bag sequence number that record sent at that time, the byte number that has sent and timestamp at that time are by the RTP bag sequence number of front and back twice RTCP bag corresponding record, sent the size that difference between the byte number information obtains average each RTP bag.
Further, said method also can have following characteristics: in the described step (b), if the report of described RTCP bag has packet drop and present packet loss to go beyond the limit, choose earlier with the employed data link bandwidth of the immediate user of described target bandwidth as the correction target bandwidth, from the medium of different encoded bandwidth, choose again and the immediate media content of described correction target bandwidth sends.
In sum, the RTP bag that the RTCP that the present invention returns by integrated treatment wraps and sends can be located fast and need be reduced to great bandwidth, avoids random adjustment medium bandwidth, simultaneously can also adjust bandwidth in advance, avoid the too high duration of encoded bandwidth oversize and packet loss occurs.Under the situation of network bandwidth fluctuation, guaranteed the media play effect.The adaptive method of realization dynamic bandwidth proposed by the invention was compared with former implementation method, have bandwidth and adjust efficient height and the strong characteristics of exploitativeness, also can carry out the adjustment of medium bandwidth in advance simultaneously, just do not carry out the adjustment of medium bandwidth and do not need to occur by the time packet loss, guaranteed user's streaming media playing effect.
Description of drawings
Fig. 1 is the adaptive applied environment of embodiment of the invention dynamic bandwidth;
Fig. 2 is that stream media system receives the dynamic bandwidth adaptation processing flow process of carrying out behind the RTCP bag that terminal returns at every turn in the embodiment of the invention.
Embodiment
Describe the inventive method in detail below in conjunction with embodiment and accompanying drawing, the streaming media playing terminal sends the RTCP bag according to the regulation of RFC1889 to streaming server in the inventive method.
Present embodiment adopts the Streaming Media under the wireless network environment to be applied as example, and the applied environment of whole bandwidth adaptation becomes as shown in Figure 1.Stream media system is by Bandwidth estimation module, and RTCP wraps receiver module, and RTP wraps sending module, and bandwidth is adjusted judge module, and medium select the media content of module and different bandwidth to form, and only list the relevant module of describing with the present invention of method here.Wherein:
The RTCP receiver module is handled the RTCP bag that receives, and judges legitimacy and the overtime situation of RTCP.
Bandwidth estimation module is the RTCP package informatin by collecting mainly, the bandwidth that estimates transmission bandwidth and need to switch.
Bandwidth is adjusted judge module by judging whether to reach the bandwidth regularization condition, and definite target is adjusted amount of bandwidth.
Medium are selected module to adjust bandwidth according to target and are selected the approaching media content of bandwidth.Wherein code rate 1, code rate 2, code rate 3 are represented the media content of different coding bandwidth, and medium select module to need can guarantee to switch under three kinds of bandwidth under the identical situation of content.For demand (telecommunication) service, the media content of the different bandwidth that the content representation of these three bandwidth encodes in advance can be placed in the file, also can be used as three files and deposits respectively.For live broadcast service, this is the real-time media stream of three different bandwidths.
Mobile phone terminal is set up data by the mobile gateway of wireless network with stream media system and is linked, under the normal condition, mobile phone terminal adopts RTSP (real-time Transmission session protocol) protocol conversation and stream media system to set up Streaming Media control channel, after the mobile phone terminal request media play, stream media system adopts the RTP bag to send to mobile phone terminal Media Stream.Mobile phone terminal at first carries out the buffering of a period of time, begins media play then.In the media play process, mobile phone terminal sends RTCPRR (receiver report, i.e. Receiver Report) bag according to certain rule to stream media system, wherein include the accumulative total number of dropped packets, receive the maximum sequence number of bag, present packet loss parameter, delay parameter.When transmission bandwidth changed, stream media system selected the media content of appropriate bandwidth to send to mobile phone terminal, guarantees the impression that user media is play.Below implementation method is described in detail.
After mobile phone terminal and stream media system are set up the stream protocol session, selected initial medium amount of bandwidth and record, select the media content of corresponding encoded bandwidth to send then, the encoded bandwidth of initially choosing is the high bandwidth of this transmission course, and after this being worth with this when adding the media giant bandwidth is the upper limit; In process of transmitting, stream media system is the RTCP bag that returns of receiving terminal simultaneously, carries out the adaptive processing procedure of following dynamic bandwidth after receiving the RTCP bag that terminal returns at every turn, as shown in Figure 2, may further comprise the steps:
Step 110, stream media system calculates the maximum duration interval that receives next RTCP bag and timer is set according to the RTCP bag of receiving, if also do not receive the RTCP bag this moment, can set this maximum duration at interval by empirical value;
In the present embodiment, when the RTCP of stream service system receiver module receives a RTCP RR bag, write down its time of advent of T, accumulative total packet loss number loss, receive the maximum sequence number RtcpSequ of bag, present packet loss parameter, and delay parameter adopt the RTCP interval calculation method of RFC1889 regulation to calculate the RTCP RR bag time interval.In the present embodiment, RTCP is wrapped 2 times of values that maximum duration is set at the above-mentioned RTCP RR bag time interval that calculates at interval.In another embodiment, above-mentioned RTCP bag maximum duration can directly adopt the test empirical value at interval.
Step 120, stream media system judge whether this RTCP wraps is to receive in the RTCP bag maximum duration interval of setting when a RTCP wraps on receiving, if carry out step 140; Otherwise RTCP wraps receive time-out, thinks the RTCP packet loss, and network condition is very poor, needs to adjust the medium bandwidth that is transmitted, and carry out step 130;
When just beginning to send, can rule of thumb be worth and an initial RTCP bag maximum duration is set at interval, judge whether overtime with this time interval.
Step 130, stream media system judge whether to exist the media content of low level encoder bandwidth, if having, select the media content of low level encoder bandwidth to send, and processing this time finishes; Otherwise, stop media content and send, finish;
Step 140, stream media system judge whether the RTCP bag that receives comprises the RR bag, if, carry out step 150, otherwise, represent that complete packet loss takes place, and carry out step 130;
Step 150 judges whether the RTCP RR bag that receives belongs to the bag that current transmission bandwidth sends, if carry out step 160, otherwise processing this time finishes;
When needing each switching media bandwidth, streaming media server medium sending module is noted initial RTP bag sequence number value, be used for judging whether the reception bag of RTCP bag statistics belongs to current bandwidth, if the reception bag more than 80% is the bag that current bandwidth sends, think that then this RTCP bag is for effectively wrapping, otherwise think invalid RTCP bag, no longer carry out follow-up dynamic bandwidth adaptation processing.
Step 160 is calculated the mean size that each RTP that this section send in the period wraps according to the record to this RTCP bag and last RTCP bag;
Stream service system whenever receives a RTCP bag, just the RTP bag sequence number that sent at that time of record, the byte number that has sent and timestamp at that time, when receiving that next RTCP wraps, the RTP bag sequence number that same record sent at that time, send byte number and timestamp at that time, by the RTP bag sequence number of front and back twice RTCP bag corresponding record, sent the size that difference between the byte number information can obtain average each RTP bag.
Step 170, the difference of the highest serial number by adjacent two RTCP bag, accumulation number of dropped packets difference obtains the RTP bag number that streaming media playing terminal actual reception arrives, and combines with the size of average each bag, obtains the byte number that the streaming media playing terminal receives;
Step 180 according to the real time interval of this RTCP bag of reception and last RTCP bag and the byte number that stream media terminal receives, estimates transmission bandwidth and record;
Wrap pairing timestamp from continuous two RTCP that streaming server write down, calculate the time interval of RTCP bag.
Step 190 is averaged as the target bandwidth to preceding estimation bandwidth several times, adopt in this example preceding twice, the mean value of three estimated values altogether;
Step 200 judges whether that the report of RTCP bag has packet drop and present packet loss to go beyond the limit, if expression needs to adjust the medium bandwidth that is sent out, and carry out step 210, otherwise carry out step 230;
Above-mentioned packet loss is to judge according to the fraction lost parameter in the RTCP RR bag, in the present embodiment, above-mentioned packet loss goes beyond the limit and refers to the packet loss parameter greater than 5, in concrete enforcement, can select other numerical value according to actual conditions, scope can be between 5 to 255.
Step 210 is according to employed data link bandwidth of user and above-mentioned target bandwidth, the target bandwidth that obtains proofreading and correct;
Because under the cdma wireless environment, the multiple that the employed data link bandwidth of user is 9.6Kbps for other standard network, can adopt other value to replace 9.6Kbps.A desirable value with the immediate 9.6Kbps multiple of target bandwidth is as the correction target bandwidth.
Step 220, from the medium of different encoded bandwidth, choose with the immediate media content conduct of correction target bandwidth and send, the encoded bandwidth media content of the difference absolute value minimum of correction target bandwidth therewith just, finish the processing of adjusting the medium bandwidth after, processing this time finishes;
When carrying out media coding, also need encode at common bandwidth situation, be convenient to carry out above-mentioned selection.
Step 230, whether the medium bandwidth of judging above-mentioned target bandwidth and current transmission basically identical, if carry out step 240; Otherwise the medium bandwidth of above-mentioned target bandwidth and current transmission differs bigger, need reselect medium bandwidth to be sent, carry out step 210;
Whether the medium bandwidth of judging target bandwidth and current transmission in the present embodiment in the following way differs too big: under existing transmission bandwidth, if the media data in a second can not be sent to the streaming media playing terminal at the buffering area of stream media terminal in the time, think that then target bandwidth and medium bandwidth differ too big.The buffering area time of stream media terminal can be made as 2s.
Step 240, judging whether to increase bandwidth, if, the medium bandwidth is adjusted to the medium bandwidth of upper level, processing this time finishes; Otherwise do not increase the medium bandwidth, processing this time finishes.
In the present embodiment, be that used bandwidth judges whether to increase bandwidth when judging existing bandwidth whether less than initial session.
In another embodiment of the present invention, when RTCP bag receiver module receives the RTCP bag that returns at every turn, judge earlier whether the RTCP RR bag that receives belongs to the bag that current transmission bandwidth sends, if, calculate and be provided with the largest interval time of next RTCP bag again, carry out the flow process of back; Otherwise directly abandon this bag, be left intact.
Below only be preferred embodiments of the present invention, not in order to restriction the present invention, all any modifications of being done within the spirit and principles in the present invention are equal to replacement and improvement etc., all should be included within protection scope of the present invention.

Claims (10)

1, the adaptive method of a kind of dynamic bandwidth, be applied to have the stream media system of the media content of multiple different coding bandwidth, after terminal and stream media system were set up the stream protocol session, stream media system was handled when receiving RTCP Real-time Transport Control Protocol RTCP bag according to the following steps:
(a) estimate transmission bandwidth and record, again the transmission bandwidth of several times estimation is before averaged as the target bandwidth;
(b) judge whether that described RTCP bag report present packet loss goes beyond the limit, if select an encoded bandwidth media content immediate to send with it according to described target bandwidth.
2, the method for claim 1, it is characterized in that, in the described step (a), after terminal and stream media system are set up the stream protocol session, preserve the medium amount of bandwidth of choosing for the first time, and being high bandwidth with this bandwidth, in the described step (b), RTCP bag present packet loss does not go beyond the limit as described, judge that more whether basically identical and medium bandwidth also can increase for current transmission bandwidth and target bandwidth, in this way, select the media content of upper level encoded bandwidth to send, finish this time to handle.
3, method as claimed in claim 1 or 2, it is characterized in that, in the described step (b), RTCP bag present packet loss does not go beyond the limit as described, judge whether basically identical of current transmission bandwidth and target bandwidth again, if and the medium bandwidth can also increase, select the media content of upper level encoded bandwidth to send; Otherwise the medium bandwidth of described target bandwidth and current transmission differs bigger, reselects medium bandwidth to be sent, and then the medium bandwidth of judging described target bandwidth and current transmission basically identical whether.
4, the method for claim 1 is characterized in that, in the described step (a), stream media system only after receiving legal RTCP bag, is just estimated transmission bandwidth and record.
5, method as claimed in claim 4, it is characterized in that, in the described step (a), if stream media system is received the RTCP bag in RTCP bag maximum duration interval, comprise the RR bag in the described RTCP bag, and the part that the corresponding RTP of described RTCP bag wraps corresponding to the medium that transmitting surpasses the setting thresholding, and then described RTCP bag is legal RTCP bag.
6, the method for claim 1, it is characterized in that, in the described step (a), stream media system is when receiving the RTCP bag, be set the longest blanking time that receives next RTCP bag, if next RTCP wraps in to return in the longest described blanking time and comprise RR and wraps, then estimate transmission bandwidth and record; Otherwise, judge whether to have the media content that hangs down the level encoder bandwidth, if having, select the media content of low level encoder bandwidth to send, processing this time finishes; If the media content of low level encoder bandwidth does not stop whole media content and sends.
7, method as claimed in claim 5, it is characterized in that, in the described step (a), if next RTCP wraps in to return in the longest described blanking time and comprise RR and wraps, to judge also whether the corresponding RTP bag of described RTCP bag surpasses the setting thresholding corresponding to the part of the medium that transmitting, if estimate transmission bandwidth and record again; Otherwise directly finish.
8, the method for claim 1, it is characterized in that in the described step (a), stream media system is when receiving the RTCP bag, according to the size of average each the RTP bag that calculates and the RTP bag number that streaming media playing terminal actual reception arrives, obtain the byte number that the streaming media playing terminal receives; The byte number that receives according to the time interval and the stream media terminal of RTCP bag again, estimation transmission bandwidth and record.
9, as claim 1 or 7 described methods, it is characterized in that, in the described step (a), stream service system is when receiving a RTCP bag, the RTP bag sequence number that record sent at that time, the byte number that has sent and timestamp at that time are by the RTP bag sequence number of front and back twice RTCP bag corresponding record, sent the size that difference between the byte number information obtains average each RTP bag.
10, the method for claim 1, it is characterized in that, in the described step (b), if the report of described RTCP bag has packet drop and present packet loss to go beyond the limit, choose earlier with the employed data link bandwidth of the immediate user of described target bandwidth as the correction target bandwidth, from the medium of different encoded bandwidth, choose again and the immediate media content of described correction target bandwidth sends.
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WO2009106015A1 (en) * 2008-02-27 2009-09-03 华为技术有限公司 Dynamic bit rate allocation method, packet-domain streaming media server
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