CN100461906C - Method and device for improving CDMA system voice interface transmission efficiency - Google Patents
Method and device for improving CDMA system voice interface transmission efficiency Download PDFInfo
- Publication number
- CN100461906C CN100461906C CNB2006100724780A CN200610072478A CN100461906C CN 100461906 C CN100461906 C CN 100461906C CN B2006100724780 A CNB2006100724780 A CN B2006100724780A CN 200610072478 A CN200610072478 A CN 200610072478A CN 100461906 C CN100461906 C CN 100461906C
- Authority
- CN
- China
- Prior art keywords
- frame
- speech data
- speech
- interface
- data frame
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Expired - Fee Related
Links
Images
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04W—WIRELESS COMMUNICATION NETWORKS
- H04W28/00—Network traffic management; Network resource management
- H04W28/02—Traffic management, e.g. flow control or congestion control
- H04W28/06—Optimizing the usage of the radio link, e.g. header compression, information sizing, discarding information
Abstract
The method comprises: a) at sending end of voice data frame, the un-enabled voice frame is identified out from the voice data stream; b) presetting a deleting strategy to delete a part of un-enabled voice frames or overall un-enabled voice frames; sending the processed voice data frame stream to the voice interface of transmitting end. Said un-disabled voice frames are 1/8 rate frames.
Description
Technical field
The present invention relates to the mobile communication technology field, the method and apparatus of voice interface transmission efficiency in particularly a kind of raising CDMA multiplexing (CDMA) system.
Background technology
CDMA multiplexing (CDMA) system configuration as shown in Figure 1, at up direction, the signal that terminal (MS) is sent is by base station transceiver (BTS) and base station controller (BSC) or group control of functional node (PCF), connect packet data serving node (PDSN) or packet switching center (MSC), be forwarded to internet (Internet), PLMN (PLMN), public switched telephone network (PSTN) or Integrated Service Digital Network by PDSN or MSC again; At down direction, signal then sends by opposite order.For whole C DMA system, hope can improve the Abis interface between BTS and the BSC and the transmission bandwidth of the A interface between BTS and the MSC (comprising A1 interface, A2 interface and A5 interface) as much as possible, promptly interface can the voice-bearer data capacity.This just need improve the efficiency of transmission of above-mentioned interface as much as possible under the certain situation of the hardware condition of the transmission equipment that carries Abis interface and A interface.
The data of being transmitted in cdma system are the speech data frames that obtain by behind certain speech coding algorithm, a speech data frame comprises frame head and speech payload two parts, wherein frame head is for realizing that speech data frame transmits the frame format information that defines in interface and physical layer, and speech payload is the data to forming behind the acoustic coding.The main voice coding/decoding algorithms that uses following three kinds of variable bit rates in CDMA2000:
Enhancement mode variable Rate audio coder ﹠ decoder (codec) (Enhanced Variable Rate Codec, EVRC), high pass Code Excited Linear Prediction 13K audio coder ﹠ decoder (codec) (Qualcomm Code Excited LinearPrediction 13K, QCELP13K) and high pass Code Excited Linear Prediction 8K audio coder ﹠ decoder (codec) (Qualcomm Code Excited Linear Prediction 8K, QCELP8K).The speech data frame length of different code decode algorithms under various speed and the speed of eating dishes without rice or wine are as shown in table 1:
Table 1
Since in the conversation always a side is saying, the opposing party tin, and there is the part space in the gap of speaking, so can be divided into efficient voice and background noise two parts to the sound that transmits in the communication process.In above-mentioned encryption algorithm, the speech payload of full rate is the coding to efficient voice, and the speech payload of half rate and 1/4 rate frame is the coding to efficient voice and background noise transition band, the speech frame of above-mentioned these speed voice activated frame that is otherwise known as; 1/8 frame is the coding to background noise, and non-voice activated frame is otherwise known as.
In the prior art, speech frame is in the process by Abis interface or the transmission of A interface, and non-voice activated frame and voice activated frame are transmitted together.Because in fact non-voice activated frame do not transmit efficient voice, therefore this shared a part of transmission bandwidth of non-voice activated frame, actual is waste to speech interface valuable bandwidth resource.
Summary of the invention
In view of this, the objective of the invention is to, propose a kind of method that improves the cdma system voice interface transmission efficiency, can improve the cdma system voice interface transmission efficiency, avoid the waste of bandwidth.
This method comprises the steps:
A, at the transmitting terminal of speech data frame, identify the non-voice activated frame in the audio data stream;
B, according to the deletion strategy that sets in advance, the non-voice activated frame that deletion is identified sends to the audio data stream after handling the speech interface of transmitting terminal.
The described non-voice activated frame that identifies in the audio data stream of steps A is: the current speech data frame that will send is carried out physical layer rate judge that if be 1/8 speed, then this speech data frame is non-voice activated frame.
The identified non-voice activated frame of the described deletion of step B is: delete identifiedly, and frame number is the non-voice activated frame of even number.
Preferably, after step B, this method further comprises:
C, at the receiving terminal of speech data frame, identify quiet in the audio data stream;
D, the non-voice activated frame of generation also are filled in the quiet position of being identified.
At the receiving terminal of speech data frame a duration being set is that speech data frame sends timer at interval, and step C is described to be identified quiet in the audio data stream and further comprise:
C1, when receiving a speech data frame, then set timer picks up counting, if received next speech data frame before overtime, then this timer is resetted and picks up counting again; If timer expiry, then this timer this pick up counting overtime between during this period of time for quiet.
The non-voice activated frame of the described generation of step D is:
Generate speech payload and be encoded to FF, the speech data frame number is 1/8 rate frame that a last voice frame number adds 1 mould 16.
Described speech interface is Abis interface or A interface.
The present invention also aims to, propose a kind of device that improves the cdma system voice interface transmission efficiency, can improve the cdma system voice interface transmission efficiency, avoid the waste of bandwidth.
This device is arranged in the equipment that is connected with speech interface, and is connected with the interface processing module with the channel processing module of this equipment, comprises as lower module:
The rate identification unit is used to receive from the speech data frame of channel processing module and with the speech data frame that is received and sends to the deletion processing unit; Judge the whether voice activated frame of right and wrong of the speech data frame received,, and the deletion index signal that is generated is sent to the deletion processing unit if then generate the deletion index signal;
The deletion processing unit is used to preserve the deletion processing policy that sets in advance; According to deletion index signal that is received and deletion processing policy the speech data frame of current reception is carried out deletion action; Perhaps the speech data frame that is received is sent to the interface processing module.
Preferably, this device further comprises:
Quiet time identifying unit is used to receive from the speech data frame of interface processing module and with the speech data frame that is received and sends to recovery unit; Judge whether to have occurred quiet, recover index signal if then send one to recovery unit;
Recovery unit, the speech data frame that is used for being received sends to channel processing module; When receiving the recovery index signal, then generate a non-voice activated frame frame and send to channel processing module.
Described non-voice activated frame is 1/8 rate frame.
The described equipment that is connected with speech interface is base station transceiver, base station controller or mobile switching centre.
As can be seen from the above technical solutions, at the transmitting terminal of speech data frame, identification is also deleted non-voice activated frame according to certain strategy; At the receiving terminal of speech data frame, the quiet place that forms at the non-voice activated frame of deletion fills out non-voice activated frame; Thereby make speech interface transmit the decreased number of non-voice activated frame, improve the ability of speech interface voice-bearer data.
Description of drawings
Fig. 1 is the structural representation of prior art cdma system;
Fig. 2 is the frame structure schematic diagram of Abis interface speech frame;
Fig. 3 is an embodiment of the invention system schematic;
Fig. 4 is an embodiment of the invention device schematic diagram;
Fig. 5 is an embodiment of the invention process chart.
Embodiment
Core content of the present invention is: at the transmitting terminal of speech frame, the speech frame that is sent is discerned, identify non-voice activated frame wherein,, again speech frame is sent to the transmission channel of speech interface according to certain strategy deletion or whole non-voice activated frames.In order to guarantee that audio data stream is continuous, the present invention program also is included in the receiving terminal of speech data frame, detects quiet in the data flow that the speech frame that received forms, at the non-voice activated frame of quiet position compensation.
For making the purpose, technical solutions and advantages of the present invention clearer, the present invention is further elaborated below in conjunction with the drawings and specific embodiments.
The described speech interface of present embodiment is the Abis interface between BTS and the BSC.The protocol stack of Abis interface transmission as shown in Figure 2, the speech payload of speech frame partly is the speech data after handling by certain coding method, described coding method can be a kind of among above-mentioned EVRC, QCELP13K or the QCELP8K; The frame header branch comprises Abis frame head and transmission bearer frame head, and wherein the Abis frame head is used to guarantee the Abis interface reliable transmission of speech data between BTS and BSC, and the transmission bearer frame head is used for guaranteeing that speech data is in the physical channel reliable transmission.
Because the Abis interface of cdma system does not have standardization, the Abis header content of each producer's definition has a great difference, and the length of Abis frame head is 4～8 bits usually; The transmission bearer frame head also according to the physical channel difference of carrying, has ATM(Asynchronous Transfer Mode), IP, High-Level Data Link Control (HDLC) or self-defining form or the like, and length is generally 4～10 bits.
Can judge by physical layer rate to obtain the speed of each speech data frame, be the also voice activated frame of right and wrong of voice activated frame thereby judge this rate frame; And the frame number that has comprised speech frame in the Abis frame head can be formulated different non-voice activated frame processing policies according to frame number, and as the non-voice activated frame of whole deletions, or only to delete frame number be non-voice activated frame of even number or the like.
Before the processing of the non-voice activated frame of deletion, the audio data stream that speech frame is formed is continuous, does not have quiet appearance; And after deletion handles, just formed quietly in the deleted place of non-voice activated frame, therefore only need get final product at the non-voice activated frame of quiet position compensation.
The present embodiment system as shown in Figure 3, BTS and BSC the interface processing module by separately respectively insert Abis interface, between BTS and original channel processing module of BSC and interface processing module, add the compression processing module respectively, wherein,
The compression processing module of BTS equipment is used for identifying the non-voice activated frame of professional speech data frame, and according to the tactful deletion that sets in advance or whole non-voice activated frame; Also be used for discerning the quiet of downlink speech frame, generate non-voice activated frame and insert the quiet position of being identified.
The compression processing module of BSC equipment is used for analyzing the characteristic information of downlink speech frame Abis frame head, identifies non-voice activated frame, and according to preassigned tactful deletion or whole non-voice activated frame; Also be used for discerning the quiet of ascending voice frame, generate non-voice activated frame and insert the quiet position of being identified.
As can be seen from the above description, function in BTS compression processing module that increases and the compression processing module that increases at BSC is identical, with BTS equipment is example, the composition, annexation of the internal element of compression processing module and with the annexation of outside other modules as shown in Figure 4:
Rate identification unit 401, be used to receive speech data frame from the channel processing module of BTS equipment, and this speech data frame is carried out physical layer rate judge, if this speech data frame is 1/8 speed, then generate one and allow the deletion index signal, and the permission that generated deletion index signal and this speech data frame sent to deletion processing unit 402, otherwise this speech data frame is sent to deletion processing unit 402 separately.
Deletion processing unit 402 is used to preserve the deletion processing policy that sets in advance; When the speech data frame received from rate identification unit 401, judge whether to receive and allow to delete index signal, if receive, then whether decision deletes this speech data frame according to the deletion processing policy, if this speech data frame of deletion, the deletion back is quiet in one section shared of time period formation of this speech data frame; If do not receive to allow the deletion index signal or do not delete this speech data frame, then the speech data frame of being received sent to the interface processing module according to deleting the processing policy decision.
By above two unit, when can be implemented in speech interface transmission speech data frame, non-voice activated frame is deleted processing; When receiving the audio data stream of handling through deletion for being implemented in, at the non-voice activated frame of quiet position compensation, then this module also will comprise:
Quiet time identifying unit 403 is used to receive from the speech data frame of interface processing module and with the speech data frame that is received and sends to recovery unit 404; Judge whether to have occurred quiet, if then send one and recover index signal to recovery unit 404; Quiet way appears in described judgement: a timer that is used to carry out quiet time judgement is set, and this timer duration is identical at interval with the speech data frame transmission of Abis interface, is 20 milliseconds; Then this timer execution reset operation is also picked up counting again when receiving a speech data frame; If timer expiry illustrates then at timer that this begins to be timed to not receive speech frame during this period of time between overtime, promptly occurred one because the deleted mute interval that causes of 1/8 frame.
Recovery unit 404, the speech data frame that is used for being received sends to channel processing module; When receiving the recovery index signal, then generate one 1/8 frame and send to channel processing module.Described generation 1/8 frame is: generate 1/8 frame that speech payload is encoded to FF, and frame number adds 1 mould 16 by the last voice frame number in quiet position and obtains.Because frame number is 0～15 circulation, if the former frame frame number is 15, the frame number of 1/8 frame of then being inserted is 0.The speech payload coding of 1/8 frame that is generated can also be the coding of other comfort noise.
With reference to the description of above-mentioned compression processing module to BTS equipment, can obtain the composition structure and the annexation of the compression processing module of BSC equipment, so repeat no more.
Adopt said apparatus as shown in Figure 5, comprise the steps: the handling process of ascending voice frame
Step 501: to the ascending voice frame of BTS channel processing module output physical layer rate judge, identify 1/8 frame;
Step 502: according to the tactful deletion that pre-establishes or whole 1/8 frame, and the upstream voice data that will delete after handling flows the transmission channel that is sent to Abis interface by the interface processing module of BTS;
The interface processing module of step 503:BSC receives the upstream voice data stream of Abis interface, identifies quiet in the received data stream;
Step 504: generate 1/8 frame and insert the quiet position of being identified, then the ascending voice frame is sent to the subsequent treatment module of BSC.
Handling process to downlink speech frame can draw with reference to above-mentioned flow process, so repeat no more.
Use the sound effect after P.862 method is handled the embodiment of the invention to assess, the encryption algorithm of speech data is EVRC.After the embodiment of the invention was handled, the mean opinion score of speech data (MOS) was 3.739 minutes, and prior art processing back speech data must be divided into 3.798 fens.The present invention handles and causes MOS only to descend 0.059 fen, can think that speech quality does not descend.Therefore, the embodiment of the invention is not influencing under the prerequisite of speech quality substantially, has reduced the waste of Abis interface bandwidth resources, greatly improves Abis interface voice bearer efficient.If adopt the E1 interface in the Abis interface, the speech channel number of every road E1 interface bearing can be brought up to more than 240 speech channels from 180～190 common speech channels.
The present invention program can also be applied to other speech interfaces in the cdma system, for example is applied to the A interface between BSC and the MSC, comprises interfaces such as A1, A2, A5.
The above only is preferred embodiment 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, is equal to and replaces and improvement etc., all should be included within protection scope of the present invention.
Claims (12)
1. a method that improves CDMA multiplexing cdma system voice interface transmission efficiency is characterized in that, this method comprises the steps:
A, at the transmitting terminal of speech data frame, identify the non-voice activated frame in the audio data stream;
B, according to the deletion strategy that sets in advance, the non-voice activated frame that deletion is identified sends to the audio data stream after handling the speech interface of transmitting terminal.
2. method according to claim 1, it is characterized in that, the described non-voice activated frame that identifies in the audio data stream of steps A is: the current speech data frame that will send is carried out physical layer rate judge that if be 1/8 speed, then this speech data frame is non-voice activated frame.
3. method according to claim 1 is characterized in that, the identified non-voice activated frame of the described deletion of step B is: delete identifiedly, and frame number is the non-voice activated frame of even number.
4. method according to claim 1 is characterized in that, after step B, this method further comprises:
C, at the receiving terminal of speech data frame, identify quiet in the audio data stream;
D, the non-voice activated frame of generation also are filled in the quiet position of being identified.
5. method according to claim 4 is characterized in that, at the receiving terminal of speech data frame a duration being set is that speech data frame sends timer at interval, and step C is described to be identified quiet in the audio data stream and further comprise:
C1, when receiving a speech data frame, then set timer picks up counting, if received next speech data frame before overtime, then this timer is resetted and picks up counting again; If timer expiry, then this timer this pick up counting overtime between during this period of time for quiet.
6. method according to claim 4 is characterized in that, the non-voice activated frame of the described generation of step D is:
Generate speech payload and be encoded to FF, the speech data frame number is 1/8 rate frame that a last voice frame number adds 1 mould 16.
7. according to each described method of claim 1 to 6, it is characterized in that described speech interface is an Abis interface.
8. according to each described method of claim 1 to 6, it is characterized in that described speech interface is the A interface.
9. a device that improves the cdma system voice interface transmission efficiency is arranged in the equipment that is connected with speech interface, and is connected with the interface processing module with the channel processing module of this equipment, it is characterized in that this device comprises:
The rate identification unit is used to receive from the speech data frame of channel processing module and with the speech data frame that is received and sends to the deletion processing unit; Judge the whether voice activated frame of right and wrong of the speech data frame received,, and the deletion index signal that is generated is sent to the deletion processing unit if then generate the deletion index signal;
The deletion processing unit is used to preserve the deletion processing policy that sets in advance; According to deletion index signal that is received and deletion processing policy the speech data frame of current reception is carried out deletion action; Perhaps the speech data frame that is received is sent to the interface processing module.
10. device according to claim 9 is characterized in that, this device further comprises:
Quiet time identifying unit is used to receive from the speech data frame of interface processing module and with the speech data frame that is received and sends to recovery unit; Judge whether to have occurred quiet, recover index signal if then send one to recovery unit;
Recovery unit, the speech data frame that is used for being received sends to channel processing module; When receiving the recovery index signal, then generate a non-voice activated frame and send to channel processing module.
11., it is characterized in that described non-voice activated frame is 1/8 rate frame according to claim 9 or 10 described devices.
12., it is characterized in that the described equipment that is connected with speech interface is base station transceiver, base station controller or mobile switching centre according to claim 9 or 10 described devices.
Priority Applications (3)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB2006100724780A CN100461906C (en) | 2006-04-17 | 2006-04-17 | Method and device for improving CDMA system voice interface transmission efficiency |
| CNA2007800003421A CN101317480A (en) | 2006-04-17 | 2007-01-25 | Method and device for voice data transmission |
| PCT/CN2007/000283 WO2007118392A1 (en) | 2006-04-17 | 2007-01-25 | A method and device for transmitting voice data |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB2006100724780A CN100461906C (en) | 2006-04-17 | 2006-04-17 | Method and device for improving CDMA system voice interface transmission efficiency |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN1984398A CN1984398A (en) | 2007-06-20 |
| CN100461906C true CN100461906C (en) | 2009-02-11 |
Family
ID=38166537
Family Applications (2)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CNB2006100724780A Expired - Fee Related CN100461906C (en) | 2006-04-17 | 2006-04-17 | Method and device for improving CDMA system voice interface transmission efficiency |
| CNA2007800003421A Pending CN101317480A (en) | 2006-04-17 | 2007-01-25 | Method and device for voice data transmission |
Family Applications After (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CNA2007800003421A Pending CN101317480A (en) | 2006-04-17 | 2007-01-25 | Method and device for voice data transmission |
Country Status (2)
| Country | Link |
|---|---|
| CN (2) | CN100461906C (en) |
| WO (1) | WO2007118392A1 (en) |
Families Citing this family (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102137440A (en) * | 2010-12-13 | 2011-07-27 | 华为软件技术有限公司 | Method and device for realizing voice communication |
| WO2018205080A1 (en) * | 2017-05-08 | 2018-11-15 | 海能达通信股份有限公司 | Method, device and apparatus for carrying channel associated signaling during voice calling process |
| CN110650167B (en) * | 2018-06-26 | 2021-02-23 | 华为技术有限公司 | Communication method and device |
Citations (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US5870397A (en) * | 1995-07-24 | 1999-02-09 | International Business Machines Corporation | Method and a system for silence removal in a voice signal transported through a communication network |
| CN1210638A (en) * | 1996-12-09 | 1999-03-10 | 株式会社东芝 | Portable radio telephone equipment and control thereof |
| CN1237304A (en) * | 1996-09-13 | 1999-12-01 | 夸尔柯姆股份有限公司 | Conference calling system and method for wireless communications channel |
| JP2000286954A (en) * | 1999-03-08 | 2000-10-13 | Lucent Technol Inc | Method and device for selectively deleting or storing only a part of recorded voice message |
| US6324188B1 (en) * | 1997-06-12 | 2001-11-27 | Sharp Kabushiki Kaisha | Voice and data multiplexing system and recording medium having a voice and data multiplexing program recorded thereon |
| CN1679312A (en) * | 2003-04-04 | 2005-10-05 | 三菱电机株式会社 | Speed sound transmitter |
-
2006
- 2006-04-17 CN CNB2006100724780A patent/CN100461906C/en not_active Expired - Fee Related
-
2007
- 2007-01-25 WO PCT/CN2007/000283 patent/WO2007118392A1/en active Application Filing
- 2007-01-25 CN CNA2007800003421A patent/CN101317480A/en active Pending
Patent Citations (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US5870397A (en) * | 1995-07-24 | 1999-02-09 | International Business Machines Corporation | Method and a system for silence removal in a voice signal transported through a communication network |
| CN1237304A (en) * | 1996-09-13 | 1999-12-01 | 夸尔柯姆股份有限公司 | Conference calling system and method for wireless communications channel |
| CN1210638A (en) * | 1996-12-09 | 1999-03-10 | 株式会社东芝 | Portable radio telephone equipment and control thereof |
| US6324188B1 (en) * | 1997-06-12 | 2001-11-27 | Sharp Kabushiki Kaisha | Voice and data multiplexing system and recording medium having a voice and data multiplexing program recorded thereon |
| JP2000286954A (en) * | 1999-03-08 | 2000-10-13 | Lucent Technol Inc | Method and device for selectively deleting or storing only a part of recorded voice message |
| CN1679312A (en) * | 2003-04-04 | 2005-10-05 | 三菱电机株式会社 | Speed sound transmitter |
Also Published As
| Publication number | Publication date |
|---|---|
| WO2007118392A1 (en) | 2007-10-25 |
| CN1984398A (en) | 2007-06-20 |
| CN101317480A (en) | 2008-12-03 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| JP4642898B2 (en) | Method and apparatus for transmitting and receiving packet data using a preset length indicator in a mobile communication system | |
| EP1074125B1 (en) | Alternating speech and data transmission in digital communications systems | |
| JP4763682B2 (en) | Providing information on upper layer packet or frame boundary in GRE frame | |
| JP4149143B2 (en) | Signaling communication method for mobile communication system | |
| WO2004002087A3 (en) | Method and system for provision of streaming data services in an internet protocol network | |
| CN1259267A (en) | Interactive video communication over a packet data network | |
| CN104917671A (en) | Mobile terminal based audio processing method and device | |
| TW200913626A (en) | Efficient, secure digital wireless voice telephony via selective encryption | |
| CN100461906C (en) | Method and device for improving CDMA system voice interface transmission efficiency | |
| TWI381687B (en) | Apparatus and method for efficiently supporting voip in a wireless communication system | |
| CN101043759B (en) | Method for realizing data service through voice band data VBD mode and system thereof | |
| CN101622711B (en) | Methods and systems for silence insertion descriptor (sid) conversion | |
| EP3185505B1 (en) | Data packet transmission processing method and device | |
| US7203637B1 (en) | Transmission of compressed information with real time requirement in a packet oriented information network | |
| CN101282284B (en) | Gateway-across dual-tone multi-frequency conveying method and apparatus | |
| CN100364345C (en) | Implementation system for passing through speech, and implementation method for passing through speech | |
| US7149198B2 (en) | System and method for voice processing and transporting in a protocol independent tandem free operation manner | |
| CN100372397C (en) | Data process method of media access control protocol layer in digital trunking system | |
| EP1475929B1 (en) | Control component removing encoded frames from isochronous telecommunication stream | |
| EP2129051B1 (en) | A method and system for realizing the voice compensation in the mobile communication network | |
| WO2005034447A1 (en) | Method for processing encoded data in interconnecting different types of communication networks, and gateway apparatus | |
| RU2426250C2 (en) | Method and system for speech compensation in mobile communication system | |
| JP2000349824A (en) | Transmission and reception system for voice data | |
| US20050195861A1 (en) | Sound communication system and mobile station | |
| CN101272217A (en) | Method and device for processing audio/video signal |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| C06 | Publication | ||
| PB01 | Publication | ||
| C10 | Entry into substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant | ||
| CF01 | Termination of patent right due to non-payment of annual fee | ||
| CF01 | Termination of patent right due to non-payment of annual fee |
Granted publication date: 20090211 |