CN103152497A - Method, device and system for realizing variable bit rate in mobile phone VoIP (Voice over Internet Protocol) system - Google Patents
Method, device and system for realizing variable bit rate in mobile phone VoIP (Voice over Internet Protocol) system Download PDFInfo
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Abstract
The invention discloses a method, a device and a system for realizing a variable bit rate in a mobile phone VoIP (Voice over Internet Protocol) system, which are used for introducing the variable bit rate into the mobile phone VoIP system. The mobile phone VoIP system comprises a server. The method comprises the following steps of: determining the current network channel capability of a session taken as a unit; and judging whether the network channel capability in the session is converted or not, if so, corresponding adjusting the encoding bit rate or encoding level, and issuing the converted encoding bit rate or encoding level to all clients in the session for performing corresponding adjustment, wherein the clients are used for receiving messages carrying encoding bit rates or encoding levels issued by the server, and adjusting own encoding bit rates dynamically according to the messages.
Description
Technical field
The present invention relates to communication technical field, relate in particular to implementation method, the Apparatus and system of dynamic bit rate in a kind of mobile phone VoIP system.
Background technology
VBR(Variable Bit Rate, dynamic bit rate), when encoding exactly, the bit rate of output is revocable, it comes across the earliest audio coding software complexity according to voice data when coding and immediately determines to use which kind of bit rate.Adopt the audio coding software of dynamic bit rate can detect voice data when coding, as in low segment, the complexity of voice data is lower, encode with lower bit, in high segment, the complexity of voice data is higher, encodes with higher bit rate.Compare with traditional static bit rates, dynamic bit rate can guarantee the quality of voice, can control again the size of output file.
In mobile communcations system, the network bandwidth difference of considering various access networks is larger, as common WIFI(Wireless Fidelity, Wireless Fidelity) its bandwidth of network reaches 1Mbps or higher usually, and common GPRS network bandwidth only has tens kbps; Therefore, be necessary the Protocol at mobile phone VoIP(Voice OverInternet, the agreement of transferring voice on the internet) introduce dynamic bit rate in system, immediately to determine to use which kind of coding bit rate according to network of network situation of living in.
Summary of the invention
Main purpose of the present invention is to disclose implementation method, the Apparatus and system of dynamic bit rate in a kind of mobile phone VoIP system, to introduce dynamic bit rate in the mobile phone VoIP system.
For reaching above-mentioned purpose, the embodiment of the present invention discloses the implementation method of dynamic bit rate in a kind of mobile phone VoIP system, is applied to server side, comprising:
Take session as unit, determine the network channel ability that this session is current;
Whether the network channel ability in this session that judges conversion occurs, if so, and corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it.
In client, in the mobile phone VoIP system, the implementation method of dynamic bit rate comprises:
The message of carrying coding bit rate or coding grade that reception server issues;
Dynamically adjust the coding bit rate of self according to this message.
The present invention also discloses a kind of server, for the treatment of mobile phone VoIP business, comprising:
Network channel ability determination module is used for take session as unit, determines the network channel ability that this session is current; And
Coding control module, be used for judging according to the result of described network channel ability determination module whether the network channel ability of this session conversion occurs, if, corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it.
The present invention also discloses a kind of client, comprising:
Receiver module is used for the message of carrying coding bit rate or coding grade that reception server issues;
The dynamic code rate Executive Module is used for dynamically adjusting according to this message that above-mentioned receiver module receives the coding bit rate of self.
The present invention also discloses a kind of mobile phone VoIP system, comprising:
Server is used for take session as unit, determines the network channel ability that this session is current; Whether the network channel ability in this session that judges conversion occurs, if so, and corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it; And
Client is used for the message of carrying coding bit rate or coding grade that reception server issues, and dynamically adjusts the coding bit rate of self according to this message.
The embodiment of the present invention is introduced dynamic bit rate in the mobile phone VoIP system, be not the complexity according to voice data, but immediately determines to use which kind of coding bit rate according to network of network situation of living in.Make: when the network channel ability was higher, system used higher bit rate to encode, and high-quality voice call is provided; When the network channel ability was low, system used lower bit rate to encode, and reduces the quality of voice, guarantees low time delay and the fluency conversed.
Description of drawings
Fig. 1 is the disclosed mobile phone VoIP system of embodiment of the present invention frame diagram;
Fig. 2 is the implementation method flow chart of dynamic bit rate in Fig. 1;
Fig. 3 is the frame diagram of the disclosed server of the embodiment of the present invention;
Fig. 4 is the frame diagram of the disclosed client of the embodiment of the present invention.
Embodiment
Below in conjunction with Figure of description, specific implementation of the present invention is done a detailed description.
Embodiment one
The present embodiment discloses the implementation method of dynamic bit rate in a kind of mobile phone VoIP system.As shown in Figure 1, this mobile phone VoIP system comprises client 1 and server 2.
Server is used for take session as unit, determines the network channel ability that this session is current; Whether the network channel ability in this session that judges conversion occurs, if so, and corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it; And
Client is used for the message of carrying coding bit rate or coding grade that reception server issues, and dynamically adjusts the coding bit rate of self according to this message.
The implementation method of this dynamic bit rate as shown in Figure 2, comprises the following steps S1～step S4.
Step S1, server determine take session as unit the network channel ability that this session is current.
In the present embodiment, the main performance assessment criteria of the ability of network channel refers to network delay.In other embodiments, its performance assessment criteria also can be separately or is considered a kind of or combination in any in the factors such as network type, signal power.
This step is determined the network channel ability by server.This mainly contains following some reason:
1, simplify the processing of client, 2, network channel ability algorithm is optimized after being convenient to, because upgrade server is more more convenient than updating client, 3, client can only detect the network delay of self, and need to consider the network condition of all clients that participate in this session to the network channel ability that defines whole session.
In above-mentioned steps S1, determine that the current network channel ability of this session includes but not limited to following step S11～step S12.
Step S11, measure the network delay of the single hop passage between server and client in this session, and then draw the network channel ability of this session single hop; This client that carries that perhaps sends from the client that receives sends the needed time of packet of certain-length by calculating and/or checks that the number that receives packet buffer queue to obtain in the information of determined network channel ability the network channel ability of this session single hop; And
Step S12, comprehensively itself and the network channel ability that participates between all clients of this session are determined the network channel ability that this session is current.
Step S2, server judge whether the network channel ability in this session conversion occurs, if, corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it.
In this step, if the pre-stored mapping table that network channel ability and coding bit rate are arranged of server, the coding bit rate that can preferably directly will adjust by tabling look-up is handed down to client carries out following step S3 and step S4 for it.Otherwise, if the mapping table of network channel ability and coding bit rate is stored in client, server can directly be handed down to client with the coding grade, because encode grade and network hierarchy are all one to one usually, also be equal to so issue the coding grade network channel ability rating that issues whole session.
Extensively understand the technical scheme of the present embodiment for ease of the public, about above-mentioned steps S1 and step S2, details are as follows for example:
Average delay in server computing client end 10 seconds, according to the time-delay situation, network is divided into four grades, the averaging network time-delay is less than 180ms(millisecond, millisecond) be the first order, the averaging network time-delay is the second level greater than 180ms less than 300ms, the averaging network time-delay is the third level greater than 300ms less than 400ms, and the averaging network time-delay is the fourth stage greater than 400ms.
Need to consider all conversation participants' network condition to network classification, may be different because participate in each member's network condition of conversation, for example 3G network user A and 2G network user B conversation, server is received the heartbeat detection message that A and B client are sent, suppose that the time-delay of A and B averaging network is respectively 120ms and 350ms, the network hierarchy of A is the first order, and the network hierarchy of B is the third level.A is different from the network hierarchy of B, and the code rate of employing is also different, and the A network condition is better, adopts higher bit rate to encode, and may exceed the load of B network.Not only to consider self bandwidth of network situation when therefore client is encoded, also to consider the receiving ability of recipient's network, the as above situation in example, server should all be judged to the third level with A and B network hierarchy, and the code rate that adopts of A can not exceed the load of B network.In MPTY, server should relatively participate in the network condition of all clients of conversing, the network hierarchy that network condition is the poorest is judged to be the network hierarchy of all clients, and all clients client that Adoption Network is the poorest bit rate that can bear is encoded like this.Server responds by heartbeat detection and sends to client after client network is judged grade.
The message of carrying coding bit rate or coding grade that step S3, client server issue.
The message that step S4, client are received according to step S3 is dynamically adjusted the coding bit rate of self.During concrete operations, if the network channel ability changes, can first original encoder be turned off when next frame is encoded, then encode with new coding bit rate establishment encoder.
In this step S4, there are two kinds of methods can change the coding bit rate of encoder: to change sample frequency or voice are carried out in various degree compression.Also can adopt the method for both combinations to adjust code rate, the code rate scope is 8～30kbps(Bits Per Second, bytes per second).If server is divided into four grades with network, different grades adopts different code rates, thereby code rate can be selected four representative value: 30kbps, 24kbps, 16kbps and 8kbps, the code rate of using as four kinds of network hierarchies respectively accordingly.For example ought network state be detected fine, adopt the code rate of 30kbps, optimum voice quality is provided, and if network state is very poor, adopt the code rate of 8kbps, reduce voice quality, reduce network load, guarantee low time delay and the smoothness of conversation.
What need supplementary notes is: can configure the initial value of the code rate of customer end adopted in server profile, because network condition before talkthrough is unknown, to code rate selections of compromising, employing 16kbps.The averaging network time-delay wouldn't be calculated less than 5 times in after talkthrough initial 5 seconds, the heartbeat detection number of times that server is received, directly network is judged to be the first order, but need indicates simultaneously the client wouldn't switch code rate.Above-mentioned 3 critical value 180ms, 300ms, 400ms are through repeatedly testing the empirical value that draws in the diverse network situation.Be for example under the network of 150ms at average delay, the voice subjective feeling that adopts the 30kbps code rate to encode is better than adopting other code rate, under the network of average delay 200ms, the voice subjective feeling that adopts the 24kbps code rate to encode is optimum, and at the network of averaging network time-delay between 150ms and 200ms, adopt 30kbps and the 24kbps code rate subjective feeling of encoding similar, critical value is made as 180ms comparatively reasonable.If follow-up further test finds that critical value arranges to such an extent that be inaccurate, record the voice subjective feeling that adopts the 8kbps code rate to encode during such as average delay 350ms optimum, can reconfigure critical value adjustment by configuration file.
What need supplementary notes is, client can also initiatively be measured the network delay between itself and server, perhaps sends the needed time of packet of certain-length by calculating and/or checks that the number that receives packet in buffer queue determines the network channel ability of single hop between itself and server; Then the network channel ability information with described single hop sends server to for its execution above-mentioned steps S11.
To sum up, the present embodiment is introduced dynamic bit rate in the mobile phone VoIP system, is not the complexity according to voice data, but immediately determines to use which kind of coding bit rate according to network of network situation of living in.Thereby make: when the network channel ability was higher, system used higher bit rate to encode, and high-quality voice call is provided; When the network channel ability was low, system used lower bit rate to encode, and reduces the quality of voice, guarantees low time delay and the fluency conversed.
Embodiment two
Corresponding with above-described embodiment one, the present embodiment also provides the device of dynamic bit rate in a kind of mobile phone VoIP system.This device comprises the server and client side.
The block diagram of this server comprises network channel ability determination module 21 and coding control module 22 as shown in Figure 3.Wherein:
Network channel ability determination module is used for take session as unit, determines the network channel ability that this session is current.
Coding control module, be used for judging according to the result of described network channel ability determination module whether the network channel ability of this session conversion occurs, if, corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it.
Optionally, as shown in Figure 3, above-mentioned network channel ability determination module comprises following single hop network channel ability acquiring unit 211 and comprehensively defines unit 212.The function of each unit is described below:
Single hop network channel ability acquiring unit is used for measuring the network delay of the single hop passage between this session server and client, and then draws the network channel ability of this session single hop; This client that carries that perhaps sends from the client that receives sends the needed time of packet of certain-length by calculating and/or checks that the number that receives packet buffer queue to obtain in the information of determined network channel ability the network channel ability of this session single hop.
Comprehensively define the unit, be used for determining with the network channel ability between all clients of this session of participation the network channel ability that this session is current by comprehensive its of single hop network channel ability acquiring unit.
As shown in Figure 4, the client that matches with server comprises:
Dynamic code rate Executive Module 12 is used for dynamically adjusting according to this message that above-mentioned receiver module receives the coding bit rate of self.
Optionally, above-mentioned client also comprises:
The detection module (not shown), be used for to measure the network delay between itself and server, perhaps send the needed time of packet of certain-length by calculating and/or check that the number that receives packet in buffer queue determines the network channel ability of single hop between itself and server; And
The sending module (not shown), the network channel ability information that is used for the single hop that described detection module is definite sends server to.
To sum up, implementation method, the Apparatus and system of dynamic bit rate in the disclosed mobile phone VoIP system of the embodiment of the present invention, by introduce dynamic bit rate in the mobile phone VoIP system, not the complexity according to voice data, but immediately determine to use which kind of coding bit rate according to network of network situation of living in.Make: when the network channel ability was higher, system used higher bit rate to encode, and high-quality voice call is provided; When the network channel ability was low, system used lower bit rate to encode, and reduces the quality of voice, guarantees low time delay and the fluency conversed.
Above disclosed be only several specific embodiment of the present invention, still, the present invention is not limited thereto, the changes that any person skilled in the art can think of all should fall into protection scope of the present invention.
Claims (10)
1. the implementation method of dynamic bit rate in a mobile phone VoIP system, be applied to server side, it is characterized in that, comprising:
Take session as unit, determine the network channel ability that this session is current;
Whether the network channel ability in this session that judges conversion occurs, if so, and corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it.
2. the implementation method of dynamic bit rate in mobile phone VoIP system according to claim 1, is characterized in that, described definite current network channel ability of this session comprises:
Measure the network delay of the single hop passage between server and client in this session, and then draw the network channel ability of this session single hop; This client that carries that perhaps sends from the client that receives sends the needed time of packet of certain-length by calculating and/or checks that the number that receives packet buffer queue to obtain in the information of determined network channel ability the network channel ability of this session single hop; And
Comprehensive its determines with the network channel ability between all clients of this session of participation the network channel ability that this session is current.
3. the implementation method of dynamic bit rate in a mobile phone VoIP system, be applied to client, it is characterized in that, comprising:
The message of carrying coding bit rate or coding grade that reception server issues;
Dynamically adjust the coding bit rate of self according to this message.
4. the implementation method of dynamic bit rate in mobile phone VoIP system according to claim 3, is characterized in that, also comprises:
Measure the network delay between itself and server, perhaps send the needed time of packet of certain-length by calculating and/or check that the number that receives packet in buffer queue determines the network channel ability of single hop between itself and server;
Send the network channel ability information of described single hop to server.
5. a server, for the treatment of mobile phone VoIP business, is characterized in that, comprising:
Network channel ability determination module is used for take session as unit, determines the network channel ability that this session is current; And
Coding control module, be used for judging according to the result of described network channel ability determination module whether the network channel ability of this session conversion occurs, if, corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it.
6. server according to claim 5, is characterized in that, described network channel ability determination module comprises:
Single hop network channel ability acquiring unit is used for measuring the network delay of the single hop passage between this session server and client, and then draws the network channel ability of this session single hop; This client that carries that perhaps sends from the client that receives sends the needed time of packet of certain-length by calculating and/or checks that the number that receives packet buffer queue to obtain in the information of determined network channel ability the network channel ability of this session single hop; And
Comprehensively define the unit, be used for determining with the network channel ability between all clients of this session of participation the network channel ability that this session is current by comprehensive its of single hop network channel ability acquiring unit.
7. a client, is characterized in that, comprising:
Receiver module is used for the message of carrying coding bit rate or coding grade that reception server issues;
The dynamic code rate Executive Module is used for dynamically adjusting according to this message that above-mentioned receiver module receives the coding bit rate of self.
8. client according to claim 7, is characterized in that, described client also comprises:
Detection module, be used for to measure the network delay between itself and server, perhaps send the needed time of packet of certain-length by calculating and/or check that the number that receives packet in buffer queue determines the network channel ability of single hop between itself and server;
Sending module, the network channel ability information that is used for the single hop that described detection module is definite sends server to.
9. a mobile phone VoIP system, is characterized in that, comprising:
Server is used for take session as unit, determines the network channel ability that this session is current; Whether the network channel ability in this session that judges conversion occurs, if so, and corresponding adjustment coding bit rate or coding grade, and all clients that in real time coding bit rate after conversion or coding grade are handed down in this session adjust accordingly for it; And
Client is used for the message of carrying coding bit rate or coding grade that reception server issues, and dynamically adjusts the coding bit rate of self according to this message.
10. mobile phone VoIP system according to claim 9, is characterized in that,
Described client, also be used for to measure the network delay between itself and server, perhaps send the needed time of packet of certain-length by calculating and/or check that the number that receives packet in buffer queue determines the network channel ability of single hop between itself and server; And send the network channel ability information of described single hop to server;
Described server also is used for measuring the network delay of the single hop passage between this session server and client, and then draws the network channel ability of this session single hop; This client that carries that perhaps sends from the client that receives sends the needed time of packet of certain-length by calculating and/or checks that the number that receives packet buffer queue to obtain in the information of determined network channel ability the network channel ability of this session single hop; Then comprehensive its determines with the network channel ability between all clients of this session of participation the network channel ability that this session is current.
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Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN104980411A (en) * | 2014-04-14 | 2015-10-14 | 腾讯科技（深圳）有限公司 | Control method, server and terminal for video call and video call system |
| CN105989844A (en) * | 2015-01-29 | 2016-10-05 | 中国移动通信集团公司 | Audio transmission adaptive method and device |
| CN113573062A (en) * | 2020-04-29 | 2021-10-29 | 华为技术有限公司 | Traffic shaping method and device and electronic equipment |
Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2003019961A1 (en) * | 2001-08-27 | 2003-03-06 | Nokia Corporation | Selecting an operational mode of a codec |
| CN1520700A (en) * | 2001-11-08 | 2004-08-11 | ������������ʽ���� | Wireless communication meethod and mobile terminal used therefor |
| CN1739255A (en) * | 2003-01-21 | 2006-02-22 | 索尼爱立信移动通讯股份有限公司 | Speed data receiver with detection of channel coding rate |
| CN101119338A (en) * | 2007-09-20 | 2008-02-06 | 腾讯科技（深圳）有限公司 | Network voice communication method, system, device and instant communication terminal |
| CN101180822A (en) * | 2005-05-25 | 2008-05-14 | 艾利森电话股份有限公司 | Enhanced voip media flow quality by adapting speech encoding based on selected modulation and coding scheme (mcs) |
-
2013
- 2013-03-29 CN CN2013101082038A patent/CN103152497A/en active Pending
Patent Citations (5)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| WO2003019961A1 (en) * | 2001-08-27 | 2003-03-06 | Nokia Corporation | Selecting an operational mode of a codec |
| CN1520700A (en) * | 2001-11-08 | 2004-08-11 | ������������ʽ���� | Wireless communication meethod and mobile terminal used therefor |
| CN1739255A (en) * | 2003-01-21 | 2006-02-22 | 索尼爱立信移动通讯股份有限公司 | Speed data receiver with detection of channel coding rate |
| CN101180822A (en) * | 2005-05-25 | 2008-05-14 | 艾利森电话股份有限公司 | Enhanced voip media flow quality by adapting speech encoding based on selected modulation and coding scheme (mcs) |
| CN101119338A (en) * | 2007-09-20 | 2008-02-06 | 腾讯科技（深圳）有限公司 | Network voice communication method, system, device and instant communication terminal |
Cited By (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN104980411A (en) * | 2014-04-14 | 2015-10-14 | 腾讯科技（深圳）有限公司 | Control method, server and terminal for video call and video call system |
| CN105989844A (en) * | 2015-01-29 | 2016-10-05 | 中国移动通信集团公司 | Audio transmission adaptive method and device |
| CN105989844B (en) * | 2015-01-29 | 2019-12-13 | 中国移动通信集团公司 | Self-adaptive method and device for audio transmission |
| CN113573062A (en) * | 2020-04-29 | 2021-10-29 | 华为技术有限公司 | Traffic shaping method and device and electronic equipment |
| WO2021218479A1 (en) * | 2020-04-29 | 2021-11-04 | 华为技术有限公司 | Traffic shaping method and apparatus, and electronic device |
| CN113573062B (en) * | 2020-04-29 | 2023-12-29 | 华为技术有限公司 | Traffic shaping method and device and electronic equipment |
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Application publication date: 20130612 |