WO2012059051A1 - 控制语音编码速率的方法、设备及系统 - Google Patents

控制语音编码速率的方法、设备及系统 Download PDF

Info

Publication number
WO2012059051A1
WO2012059051A1 PCT/CN2011/081682 CN2011081682W WO2012059051A1 WO 2012059051 A1 WO2012059051 A1 WO 2012059051A1 CN 2011081682 W CN2011081682 W CN 2011081682W WO 2012059051 A1 WO2012059051 A1 WO 2012059051A1
Authority
WO
WIPO (PCT)
Prior art keywords
call
local
voice
speed adjustment
switched
Prior art date
Application number
PCT/CN2011/081682
Other languages
English (en)
French (fr)
Inventor
朱星
郭江
柳军
严凯
Original Assignee
华为技术有限公司
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by 华为技术有限公司 filed Critical 华为技术有限公司
Priority to EP11837577.3A priority Critical patent/EP2635069A4/en
Publication of WO2012059051A1 publication Critical patent/WO2012059051A1/zh

Links

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/22—Mode decision, i.e. based on audio signal content versus external parameters
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/60—Network streaming of media packets
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/006—Networks other than PSTN/ISDN providing telephone service, e.g. Voice over Internet Protocol (VoIP), including next generation networks with a packet-switched transport layer
    • H04M7/0072—Speech codec negotiation
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W4/00—Services specially adapted for wireless communication networks; Facilities therefor
    • H04W4/18—Information format or content conversion, e.g. adaptation by the network of the transmitted or received information for the purpose of wireless delivery to users or terminals
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W36/00—Hand-off or reselection arrangements
    • H04W36/34—Reselection control
    • H04W36/36—Reselection control by user or terminal equipment
    • H04W36/362—Conditional handover

Definitions

  • the present invention relates to the field of communications, and in particular, to a method, device and system for controlling a speech coding rate. Background technique
  • TrFO Transcoder Free Operat ion, free
  • the MS participating in the call in the prior art is determined according to the quality of the air interface.
  • the speech coding rate used When the air interface quality is good, a high speech coding rate can be selected; when the air interface quality is poor, a lower speech coding rate can be selected.
  • the MS participating in the call determines the voice coding rate according to the air interface quality. Therefore, in some scenarios, the highest rate allowed by the peer end may be adopted due to the good quality of the air interface at the calling end, resulting in a high rate. The rate of voice frames cannot be sent to the peer, which causes silence.
  • both parties MS1 and MS2 use the full rate set to maintain the TrFO call before the call handover occurs.
  • the compatible half rate set code is used to maintain the TrFO call, if the MS2 adopts full speed concentration.
  • the highest coding rate is voice coded, and the beacon will not receive the high-speed speech frame sent by MS2 when the call is switched, resulting in muting. Summary of the invention
  • the embodiment of the present invention provides a method, device and system for controlling the speech coding rate.
  • the technical solution is as follows: On the one hand, a method for controlling a voice coding rate is provided, the method includes: receiving a call pre-handover notification, where the call pre-handover notification carries a call taken by a local mobile terminal MS after a call handover Voice coding type;
  • a device for controlling a voice coding rate includes: a receiving module, configured to receive a call pre-handover notification, where the call pre-handover notification carries a call adopted by the local mobile terminal MS after the call is switched.
  • Voice coding type configured to transmit a call pre-handover notification to the call pre-handover notification.
  • a setting module configured to set a speed adjustment indication according to a voice coding type used after the local MS call is switched received by the receiving module
  • a sending module configured to send the speed adjustment indication set by the setting module to the peer MS, and enable the peer MS to perform voice coding according to the speed adjustment indication, until the local MS completes the call switching.
  • a method of controlling a speech coding rate comprising:
  • a base station control device is also provided, where the base station control device includes:
  • a receiving module configured to receive a voice frame
  • a modification module configured to change the speed adjustment indication carried in the voice frame received by the receiving module to a default initial rate
  • a sending module configured to send the modified voice frame carrying the default initial rate to the corresponding mobile terminal MS, so that the corresponding MS performs voice coding according to the default initial rate.
  • a base transceiver station comprising: a receiving module, configured to receive a voice frame;
  • a modification module configured to change the speed adjustment indication carried in the voice frame received by the receiving module to a default initial rate
  • a sending module configured to send the modified voice frame carrying the default initial rate to the corresponding mobile terminal MS, so that the corresponding MS performs voice coding according to the default initial rate.
  • a system for controlling a speech coding rate comprising: a base station control device and a base transceiver station;
  • the base station control device is, for example, the base station control device; and the base transceiver station is the base transceiver station.
  • the speed adjustment indication is set according to the voice coding type used after the local MS call is switched, and the speed adjustment indication is sent to the opposite MS, so that the opposite MS performs voice coding according to the speed adjustment indication, thereby avoiding the local MS in the The mute problem caused by the inability to answer the high-rate speech frame sent by the peer MS in a short period of time when the call is switched;
  • the speed adjustment indication carried in the received voice frame is modified to the default initial rate, and is sent to The corresponding MS enables the corresponding MS to perform speech coding according to the default initial rate, thereby further avoiding the mute problem caused by the inability to answer the high-rate speech frame.
  • FIG. 1 is a flowchart of a method for controlling a speech coding rate according to Embodiment 1 of the present invention
  • FIG. 2 is a flowchart of a method for controlling a speech coding rate according to Embodiment 2 of the present invention
  • FIG. 3 is a call handover according to Embodiment 3 of the present invention; Schematic diagram of the scene;
  • FIG. 4 is a schematic diagram of a call handover scenario message interaction according to Embodiment 3 of the present invention
  • FIG. 5 is a flowchart of a method for controlling a voice coding rate according to Embodiment 3 of the present invention
  • FIG. 6 is another call provided by Embodiment 3 of the present invention. Switch the scene diagram;
  • FIG. 7 is a schematic diagram of another call handover scenario message interaction according to Embodiment 3 of the present invention
  • FIG. FIG. 8 is a flowchart of another method for controlling a voice coding rate according to Embodiment 3 of the present invention
  • FIG. 9 is a schematic diagram of a call setup scenario according to Embodiment 3 of the present invention.
  • FIG. 10 is a flowchart of still another method for controlling a voice coding rate according to Embodiment 3 of the present invention.
  • FIG. 11 is a schematic structural diagram of a device for controlling a voice coding rate according to Embodiment 4 of the present invention
  • FIG. 12 is a schematic structural diagram of a transmitting module according to Embodiment 4 of the present invention.
  • FIG. 13 is a schematic structural diagram of another apparatus for controlling a voice coding rate according to Embodiment 4 of the present invention.
  • FIG. 14 is a schematic structural diagram of a base station control device according to Embodiment 5 of the present invention.
  • FIG. 15 is a schematic structural diagram of a base transceiver station according to Embodiment 6 of the present invention.
  • FIG. 16 is a schematic structural diagram of another base transceiver station according to Embodiment 6 of the present invention
  • FIG. 17 is a schematic structural diagram of a system for controlling a voice coding rate according to Embodiment 7 of the present invention. detailed description
  • the embodiment provides a method for controlling a voice coding rate, and the process is specifically as follows:
  • the speed adjustment indication is set according to the voice coding type used after the local MS call is switched; 103: the speed adjustment indication is sent to the opposite MS, so that the opposite MS performs voice coding according to the speed adjustment indication, until the local MS Complete the call switch.
  • the BTS Base Transceiver Station
  • the BTS Base Transceiver Station
  • the voice coding type after the local MS call is switched and specifically includes:
  • the BTS determines the default initial rate after the call is switched according to the voice coding type used after the local MS call is switched, and uses the default initial rate after the call is switched as the speed adjustment indication.
  • the voice coding type used after the local MS calls the handover gives the AMR rate set used after the local MS switches.
  • the protocol specifies the initial call rate as follows: If the rate set contains 4 rates, the initial call rate is the second lowest rate; if the rate set contains 2 or 3 rates, the initial call rate is the lowest rate.
  • the initial call rate may be used as a default initial rate after the call is switched according to the manner specified in the foregoing protocol. Certainly, other determining manners may also be selected. In this embodiment, the manner of determining the default initial rate after the call is switched and the specific value determined are not limited, and the local MS can still receive the normal MS after the call is switched.
  • the voice frame is OK.
  • the speed adjustment indication is sent to the peer MS, and specifically includes:
  • the BTS receives the voice frame sent by the local MS to the peer MS before the call is switched, and the voice frame carries the speed adjustment indication determined by the local MS according to the quality of the local air interface.
  • the speed adjustment indication carried in the voice frame is modified to be the speed adjustment indication set according to the voice coding type used after the local MS call is switched, and the modified voice frame is sent to the peer control device, and the peer control device will The modified voice frame is sent to the peer MS.
  • the speed adjustment indication is sent to the peer MS, which specifically includes:
  • the silent control frame carrying the speed adjustment indication is sent to the peer control device BTS, and the peer BTS will be muted.
  • the frame is modified to be padded and sent to the peer MS.
  • the preset time can be set according to the actual situation, for example, the preset time is set to be a No_Data frame, and the padding frame may be a SID_Fi ler frame. 20 milliseconds, this embodiment does not limit the specific value of the preset time. That is, if the voice frame sent by the local MS to the peer MS before the call handover is received in the preset time, the No_Data frame carrying the speed adjustment indication is sent to the peer control device, and is controlled by the peer. The device modifies the received No_Da ta frame carrying the speed control indication to
  • the peer MS After the SID-Fi ler frame is sent to the peer MS, the peer MS performs voice coding according to the speed adjustment indication carried in the SID_Fi ller frame, thereby implementing rate adjustment.
  • the method further includes:
  • the local MS initiates a call setup or advance.
  • the line call switching is taken as an example to describe the method of controlling the speech coding rate.
  • a call setup or a call handover may be initiated by the peer MS.
  • the implementation process is the same as that of the local MS, and is not described here.
  • the method provides a speed adjustment indication according to the voice coding type used after the local MS calls the call, and sends the speed adjustment indication to the opposite MS, so that the opposite MS performs voice coding according to the speed adjustment indication.
  • the local end MS performs the voice according to the default initial rate of the local end by modifying the speed adjustment indication in the voice frame sent by the core network to the default initial rate of the local end. Encoding, which avoids the problem that the local MS cannot answer the high-rate voice frame sent by the peer MS or the peer MS cannot receive the high-rate voice frame, which provides guarantee for both parties to make normal calls.
  • Embodiment 2 Embodiment 2
  • This embodiment provides a method for controlling a voice coding rate. Referring to FIG. 2, the method is specifically as follows:
  • the receiving a voice frame for the call setup scenario includes:
  • the BCS Base Channel Controller
  • the voice frame carries the speed regulation determined by the core network according to the quality of the calling MS air interface.
  • the speed adjustment indication carried in the voice frame is modified to the default initial rate, and the voice frame carrying the default initial rate is sent to the corresponding mobile terminal MS, which specifically includes:
  • the BCS modifies the speed adjustment indication carried in the voice frame sent by the core network to the default initial rate of the MS, and sends the voice frame carrying the default initial rate of the calling MS to the calling MS, so that the calling MS follows The default initial rate of the calling MS is voice coded.
  • Receiving a voice frame for a call handover scenario specifically includes:
  • the base station transceiver station BTS receives the voice frame sent by the local MS to the peer MS before the call is handed over, and the voice frame carries the speed regulation indication determined by the local MS according to the quality of the local air interface;
  • the speed adjustment indication carried in the voice frame is modified to a default initial rate, and will be carried
  • the voice frame of the default initial rate is sent to the corresponding mobile terminal MS, which specifically includes:
  • the BTS modifies the speed adjustment indication carried in the voice frame sent by the local MS to the default initial rate after the local MS is switched, and sends the voice frame carrying the default initial rate after the local MS is switched to the peer MS. , the peer MS performs voice coding according to the default initial rate after the local MS call is switched.
  • the BTS further includes:
  • the BTS receives the call pre-switching notification sent by the base station control device, and the call pre-switching notification carries the voice coding type used after the local MS call is switched;
  • the BTS determines the default initial rate after the local MS call is switched according to the voice coding type used after the local MS call is switched;
  • the BTS modifies the speed adjustment indication carried in the voice frame sent by the local MS to the default initial rate after the local MS call is switched, which specifically includes:
  • the speed adjustment indication carried in the voice frame sent by the local MS is modified to the default initial rate after the local MS call is switched according to the voice coding type used by the local MS call switch.
  • Embodiment 3 by modifying the speed adjustment indication carried in the received voice frame to a default initial rate, and sending the same to the corresponding MS, so that the corresponding MS performs voice coding according to the default initial rate, thereby avoiding Silence caused by the inability of the MS to receive high-rate speech frames during call setup or call handover.
  • This embodiment provides a method for controlling a voice coding rate.
  • the present embodiment first provides a call handover scenario in the BSC shown in FIG. 3, and the message interaction process shown in FIG. The method of controlling the speech coding rate is described in detail.
  • both parties MS1 and MS2 are in the same BSC.
  • both MS1 and MS2 use the full rate set AMR-FR Se tl ( 12. 2kpbs, 7. 4kpbs, 5. 9kpbs, 4. 95kpbs) to establish TrFO. call.
  • a compatible half-rate set AMR-HR Se tl (7.4 kkbs, 5. 9kpbs, 4.95kpbs) is used to maintain the TrFO call. If the call is switched, the MS2 uses a coding rate of 12.2kpbs.
  • the MSI For the MSI with a maximum voice coding rate of only 7.40kpbs after the call is handed over, it will cause the high-speed voice frame sent by the MS2 to be received after switching to the new channel. Mute for a short time after switching. Mute until it continues to MS1 After the handover is completed, the MSI sends a speed adjustment indication CMR (Code Mode Reques t) to the MS2 to notify the MS2 to reduce the coding rate. After the MS2 has a voice rate lower than 12. 2 kbps, the MSI can receive the voice frame from the MS2. .
  • CMR Code Mode Reques t
  • the present embodiment provides a method for controlling the voice coding rate.
  • this embodiment provides The method flow is as follows:
  • the BTS1 receives a call pre-handover notification sent by the BSC, where the call pre-handover notification carries a voice coding type used after the MS1 call is switched;
  • the BSC can determine the voice coding type adopted by the MS1 before and after the call handover.
  • the voice coding type used before and after the MS1 call handover is switched from the full rate to the half rate, and the BSC determines that the AMR full half rate switching occurs before and after the MS1 call handover, and the voice coding type used before and after the MS1 call handover is compatible.
  • the call forwarding is notified by the pre-handover notification of the pre-handover notification, and the BTS1 can determine the voice coding rate used after the MS1 call is handed over.
  • the BSC carries the voice coding type used after the MS1 call is switched in the Pre. -Handover Not if icat ion is sent to BTS1, and then the handover command Handover Command/allocation command is sent to MS1.
  • the BTS1 sets the speed adjustment indication according to the voice coding type used after the MSI call is switched. Specifically, the embodiment does not specifically limit the manner in which the BTS1 sets the speed adjustment indication according to the voice coding type used after the MS1 call is switched. According to the setting manner of the speed regulation indication described in the first embodiment, since the voice coding type used after the MS1 call handover is AMR_HR Set l ( 7. 4kpbs, 5. 9kpbs, 4. 95kpbs), the tone set by the BTS1 is adjusted. The speed indicator shall be the lowest of the three rates of 4.95kpbs.
  • the BTS1 receives the voice frame sent by the MSI to the MS2 before the call handover, and the voice frame carries the speed adjustment indication determined by the MS1 according to the quality of the local air interface.
  • the sequence of the step 503 and the step 502 is not limited.
  • the BTS1 may first receive the voice frame sent by the MS1 to the MS2 before the call handover, and then according to the voice coding type adopted after the MS1 call is switched. Set the speed indicator.
  • the BTS1 modifies the speed adjustment indication carried in the received voice frame to a speed adjustment indication set according to the voice coding type used after the MS1 call is switched, and sends the modified voice frame to the MS2.
  • the BTS2 sends the modified voice frame to the MS2, so that the MS2 performs voice coding according to the speed adjustment indication set by the BTS1.
  • step 504 and step 505 since the BTS1 modifies the speed adjustment indication in the voice frame sent by the MS1 to the MS2 before the call handover in step 504, the speed adjustment indication set by the BTS1 in the above step 502, and the speed adjustment After indicating the default initial rate after the call is switched, the BTS2 sends the modified voice frame to the MS2 in step 505, so that the MS2 performs voice coding according to the speed adjustment indication set by the BTS1, even if the MS1 calls the half-rate after the call is switched.
  • the voice frame encoded by the MS2 according to the default initial rate after the call handover can still be received normally, thereby avoiding the short-time mute of the MS1 during the call handover, and also improving the anti-interference capability.
  • the code rate can be adjusted by the autonomous speed adjustment between the MS1 and the MS2.
  • the default initial rate after call switching is changed to a higher encoding rate, which improves voice quality. For example, change to the highest encoding rate after MS1 call switching 7. 4kpbs.
  • both parties MS1 and MS2 are in the same BSC1, and both MS1 and MS2 use AMR-FR Se t l to establish a TrFO call. Subsequent MS1 switching between BSCs (from BSC1 to BSC) will be followed by a compatible AMR_HR Set l encoding to maintain TrFO calls. If the call is switched, the MS2 uses a coding rate of 12.2kpbs. For the MS1 with the highest voice coding rate of 7.40kpbs after the call is switched, it will cause the high-speed voice frame sent by the MS2 to be received after switching to the new channel. Mute for a short time after switching. After the mute state continues until the MS1 switch is completed, the MS2 is notified by CMR to reduce the coding rate. After the MS2 has a voice rate lower than 12. 2 kbps, the MS1 can receive the voice frame from the MS2.
  • the method provided in this embodiment switches the call between the BSCs by combining the message interaction process shown in FIG.
  • the message is extended, so that the corresponding BSC1' informs the BSC1 of the voice coding type adopted after the MS1 call is switched after the MS1 call is handed over, thereby realizing the control of the voice coding rate.
  • FIG. 8 the method for controlling the voice coding rate provided by this embodiment is shown. The legal process is as follows:
  • the MSC Mobile Switching Center
  • the MS1 corresponding to the call handover occurs.
  • the BSC1 does not determine in advance which type of voice coding the MS1 uses after the call is handed over. Therefore, the method provided in this embodiment carries the voice coding type adopted after the MS1 call is switched by the BSC1' in the handover request to confirm the HANDOVER REQUEST.
  • the ACKN0WLEGE is sent to the corresponding MSC, so that after receiving the HANDOVER REQUEST ACKN0WLEGE sent by the BSC1', the MSC learns the voice coding type Codec that will be adopted after the MSI call is switched.
  • the MSC carries the voice coding type used after the MSI call is switched in the handover command and sends it to the BSC 1;
  • the MSC sends a handover command HANDOVER COMMAND to the BSC1, and by adding the coding type indication information used after the MSI call handover in the HANDOVER COMMAND, the BSC1 can learn the voice coding type used after the MS1 call is switched.
  • the embodiment does not limit the manner in which the coding type indication information used after the MSI call is switched in the HANDOVER COMMAND. For example, in the existing A-port HANDOVER C0 ⁇ AND message, the NewBSS to Old BSS Information cell can be extended, and the Speech Codec (Coussen) field is used to indicate the type of codec to be used after the MSI call is switched.
  • the BSC1 After receiving the handover command sent by the MSC, the BSC1 sends a call pre-handover notification to the BTS1, where the call pre-handover notification carries the voice coding type used after the MS1 call is switched; wherein, after receiving the HANDOVER COMMAND sent by the MSC, the BSC1 receives the HANDOVER COMMAND If the full-rate AMR encoding is used before the MS1 call is handed over, the BSC1 can learn that the half-rate channel compatible with the pre-switching AMR rate set is used after the MS1 call is handed over.
  • the BSC1 carries the voice coding type that is used after the MS1 call is switched in the Pre-Handover Notification and sends it to the BTS1.
  • the BTS1 sends the voice frame sent to the peer MS2, the speed adjustment indication in the voice frame is set to be after the call is switched.
  • the BSC1' may also carry the interface bearer type in the HANDOVER REQUEST ACKN0WLEGE sent to the MSC, and the BSC1 needs to determine whether the interface bearer type before and after the MS1 call handover is IP, and after determining the IP bearer after the handover, Send it to BTS1 again Pre-Handover Notification
  • the BSC1 can also omit the judging step, which is not specifically limited in this embodiment.
  • the BTS1 sets the speed adjustment indication according to the voice coding type used after the MS1 call is carried in the call pre-switching notification.
  • the BTS1 receives the voice frame sent by the MS1 to the MS2 before the call is handed over, and the voice frame carries the speed adjustment indication determined by the MS1 according to the quality of the local air interface.
  • the BTS1 modifies the speed adjustment indication carried in the received voice frame to a speed adjustment indication set according to the voice coding type used after the MS1 call switching, and sends the modified voice frame to the BTS2 corresponding to the MS2;
  • the BTS2 sends the modified voice frame to the MS2, so that the MS2 performs voice coding according to the speed adjustment indication set by the BTS1.
  • the steps 804 to 807 are the same as the steps 502 to 505, and the embodiment is not described herein again.
  • the inter-BSC call handover shown in FIG. 8 is a call handover performed in the same MSC.
  • the method provided in this embodiment is also supported, and the implementation manner is as shown in FIG.
  • the principle of call switching between BSCs is the same, and will not be repeated here.
  • the BTS1 sends the set speed adjustment indication to the MS2 in a manner different from the above step 504 or step 804.
  • the BTS1 carries the speed control indication set in the No_Data frame and sends it to the BTS2 corresponding to the MS2.
  • the BTS2 sends the No_data frame to the SID_Filler frame and sends it to the MS 2, so that the MS 2 can also be set according to the BTS1.
  • the speed adjustment instruction is voice coded to achieve the purpose of rate adjustment.
  • the core network when the calling parties are located in the same BSC, or in different BSCs in the same MSC, or in different MSCs, when the calling parties establish a call, if the called party does not answer the call, the core network will be triggered to the main Call the end to put back the ring tone.
  • MS1 uses the highest speech coding rate of 12.2 kbps.
  • the 12.2 kbps voice frame from the MS1 cannot be sent to the MS2 through the half-rate air interface channel, causing some high-rate voice frames to be lost.
  • the voice frame from MS2 carries the speed control indication, asking the main The MS I is adjusted to a rate of less than 12.2 kbps, and the voice frame of the calling user can be received by the called party.
  • the method provided in this embodiment further includes:
  • the calling side BSC Before the called MS answers the call, the calling side BSC receives the voice frame sent by the core network in the process of putting back the ring tone, and the voice frame carries the speed regulation determined by the core network according to the quality of the calling side air interface.
  • the speed adjustment indication carried in the voice frame sent by the core network is a higher voice coding rate, but the voice sent by the core network is sent by the calling side BSC in the subsequent steps.
  • the called MS can also receive the voice frame sent by the calling MS.
  • the calling side BSC modifies the speed control indication carried in the voice frame sent by the core network to the default initial rate of the calling side;
  • the method for determining the default initial rate of the calling side in the step may also be determined by using the default initial rate provided in the foregoing step, which is not specifically limited in this embodiment.
  • the calling side BSC1 receives the voice frame sent by the core network, that is, the MSC 1 in the process of putting back the ring tone, and the voice frame carries the MSC 1 according to the calling side.
  • the speed adjustment indication of the air interface quality is determined; the BSC1 modifies the speed adjustment indication carried in the voice frame sent by the received MSC1 to the default initial rate of the local end, and sends it to the BTS 1 , and then sends the BTS 1 to the MS 1 to make the MS 1 Perform voice coding according to the default initial rate of the local end. Since the default initial rate of the calling side where MS 1 is located is not higher than the encoding rate used by MS 2, MS 2 will be able to successfully receive the voice frame sent by MS 1, thus avoiding the silence during call setup.
  • the speed adjustment indication carried in the voice frame sent by the core network is determined according to the quality of the air interface on the calling side, the tone carried in the voice frame sent by the core network is changed according to the quality of the air interface on the calling side.
  • the speed indication is variable.
  • the core network may also choose to carry a fixed speed adjustment indication in the voice frame.
  • the speed adjustment indication carried in the voice frame sent by the core network is set as the default initial rate of the calling side. In this case, after the calling party determines that the speed adjustment indicator carried in the voice frame sent by the core network is the default initial rate of the local end, the calling party does not need to modify the voice frame sent by the core network, and can also reach the calling MS according to the local end.
  • the default initial rate is used for speech coding purposes.
  • this embodiment is This is not limited.
  • the method provides a speed adjustment indication according to the voice coding type used after the local MS calls the call, and sends the speed adjustment indication to the opposite MS, so that the opposite MS performs voice coding according to the speed adjustment indication.
  • the local end MS performs the voice according to the default initial rate of the local end by modifying the speed adjustment indication in the voice frame sent by the core network to the default initial rate of the local end. Encoding, which avoids the problem that the local MS cannot answer the high-rate voice frame sent by the peer MS or the peer MS cannot receive the high-rate voice frame, which provides guarantee for both parties to make normal calls.
  • the embodiment provides a device for controlling a voice coding rate, where the device is used to perform the method steps performed by the BTS device in the foregoing Embodiment 1 and Embodiment 3.
  • the device includes:
  • the receiving module 1101 is configured to receive a call pre-switching notification, where the call pre-switching notification carries a voice coding type used by the local mobile terminal after the MS call is switched;
  • the setting module 1102 is configured to set a speed adjustment indication according to a voice coding type adopted after the local MS call is received by the receiving module 1101;
  • the sending module 1103 is configured to send the speed adjustment indication set by the setting module 1102 to the peer MS, so that the peer MS performs voice coding according to the speed adjustment indication, until the local MS completes the call handover.
  • the setting module 1102 is specifically configured to determine a default initial rate after the call is switched according to the voice coding type used after the local MS call is switched, and use the default initial rate after the call is switched as the speed adjustment indication. For the determination of the default initial rate after the call is switched, refer to the related description of the first embodiment and the step 502 in the foregoing third embodiment, and details are not described herein again.
  • the sending module 11 03 specifically includes:
  • the receiving unit 1103a is configured to receive a voice frame that is sent by the local MS to the peer MS before the call is switched, and the voice frame carries the speed adjustment indication determined by the local MS according to the quality of the local air interface.
  • the modifying unit 1103b is configured to modify the speed adjustment indication carried in the voice frame received by the receiving unit 1103a into a speed adjustment indication set according to the voice coding type adopted after the local MS call is switched; the sending unit 1103c is configured to modify The modified voice frame of the unit 1103b is sent to the peer control device, and the modified voice frame is sent by the peer control device to the peer MS.
  • the sending module 1103 is specifically used if the preset time is The voice frame that is sent to the peer MS before the call is switched is sent to the peer control device, and the peer control device sends the silence frame to the padding frame and then sends the message to the peer control device. Peer MS.
  • the mute frame may be a No_Data frame
  • the padding frame may be a S ID_Fi l ler frame.
  • the preset time involved in this embodiment and subsequent embodiments may be set according to actual conditions, for example, setting the preset time to 20 In milliseconds, this embodiment does not limit the specific value of the preset time. That is, if the voice frame sent by the local MS to the peer MS before the call handover is received in the preset time, the No_Data frame carrying the speed adjustment indication is sent to the peer control device, and is controlled by the peer.
  • the device modifies the received No_Data frame carrying the speed control indication to the SID_Fieller frame, and then sends the signal to the peer MS, so that the peer MS performs voice coding according to the speed adjustment indication carried in the SID_Filer frame, thereby realizing the rate. Adjustment.
  • the device further includes:
  • the determining module 1104 is configured to determine whether the interface bearer type used before and after the local MS call is switched is an IP bearer.
  • the setting module 1102 is configured to: after the determining module 1104 determines that the interface carried by the local MS call is carried as an IP bearer, perform the step of setting a speed adjustment indication according to the voice coding type used after the local MS call is switched.
  • the device provided in this embodiment sets the speed adjustment indication according to the voice coding type used after the local MS calls the handover, and sends the speed adjustment indication to the opposite MS, so that the opposite MS performs voice coding according to the speed adjustment indication.
  • the local end MS performs the voice according to the default initial rate of the local end by modifying the speed adjustment indication in the voice frame sent by the core network to the default initial rate of the local end. Encoding, which avoids the problem that the local MS cannot answer the high-rate voice frame sent by the peer MS or the peer MS cannot receive the high-rate voice frame, which provides guarantee for both parties to make normal calls.
  • Embodiment 5 Embodiment 5
  • the present embodiment provides a base station control device, which is used to perform the method steps performed by the BSC in the second embodiment and the third embodiment.
  • the device includes:
  • the modification module 1402 is configured to modify the speed adjustment indication carried in the voice frame received by the receiving module 1401 to a default initial rate;
  • the sending module 1403 is configured to send the modified voice frame carrying the default initial rate modified by the modifying module 1402 to the corresponding mobile terminal MS, so that the corresponding MS performs voice encoding according to the default initial rate.
  • the receiving module 1401 is specifically configured to receive a voice frame sent by the core network in the process of releasing the ring tone, where the voice frame carries a speed adjustment indication determined by the core network according to the mass of the calling MS air interface; correspondingly, the modifying module 1402.
  • the method is specifically configured to modify, by using a voice frame sent by a core network received by the receiving module 1401, a speed adjustment indication, which is a default initial rate of the MS.
  • the sending module 1403 is specifically configured to send, by the modifying module 1402, the voice frame carrying the default initial rate of the calling MS to the calling MS, so that the calling MS performs voice encoding according to the default initial rate of the calling MS.
  • the base station control device modifies the preset speed rate of the MS to be sent to the calling MS by transmitting the speed adjustment indication carried in the received voice frame of the core network, and sends the same to the calling MS.
  • the speech coding is performed at the default initial rate, thereby avoiding the mute problem caused by the called MS being unable to answer the high-rate speech frame transmitted by the calling MS during the call setup process.
  • the present embodiment provides a base transceiver station, which is configured to perform the method steps performed by the BTS in the second embodiment and the third embodiment.
  • the base transceiver station includes: a first receiving module 1501. For receiving voice frames;
  • the modifying module 1502 is configured to modify the speed adjustment indication carried in the voice frame received by the first receiving module 1501 to a default initial rate.
  • the sending module 1503 is configured to send the modified voice frame carrying the default initial rate by the modifying module 1502 to the corresponding mobile terminal MS, so that the corresponding MS performs voice encoding according to the default initial rate.
  • the first receiving module 1501 is configured to receive a voice frame that is sent by the local MS to the peer MS before the call is switched, and the voice frame carries the speed adjustment indication determined by the local MS according to the quality of the local air interface.
  • the modifying module 1502 is specifically configured to modify the speed adjustment indication carried in the voice frame sent by the local MS received by the first receiving module 1501 to the default initial rate after the local MS call is switched;
  • the sending module 1503 is specifically configured to carry the local MS call after the modification module 1502 is modified.
  • the voice frame of the default initial rate after the handover is sent to the peer MS, so that the peer MS performs voice coding according to the default initial rate after the local MS call is switched.
  • the base transceiver station further includes:
  • the second receiving module 1504 is configured to receive a call pre-switching notification sent by the base station control device, where the call pre-switching notification carries a voice coding type used after the local MS call is switched;
  • the determining module 1505 is configured to determine, according to the voice coding type used by the local MS call switch received by the second receiving module 1504, a default initial rate after the local MS call is switched;
  • the modification module 1502 is specifically configured to modify the speed adjustment indication carried in the voice frame sent by the local MS received by the first receiving module 1501 to the default initial rate after the local MS call is determined by the determining module 1505.
  • the base transceiver station after receiving the call pre-handover notification sent by the base station control device, modifies the speed adjustment indication carried in the received voice frame sent by the local MS to the local MS after the call is switched.
  • the default initial rate is sent to the peer MS, so that the peer MS performs voice coding according to the default initial rate after the local MS calls the switch, so that the local MS cannot receive the peer MS during the call handover process.
  • This embodiment provides a system for controlling a speech coding rate.
  • the system includes: a base station control device 1701 and a base transceiver station 1702;
  • the base station control device 1701 is the base station control device provided by the foregoing embodiment 5.
  • the base transceiver station 1702 is the base transceiver station provided in the foregoing sixth embodiment.
  • the base station control device or the base transceiver station modifies the speed adjustment indication carried in the received voice frame to a default initial rate, and sends the speed adjustment indication to the corresponding MS, so that the corresponding MS follows the default initial rate.
  • Voice coding is performed to avoid the mute problem caused by the MS not being able to answer high-rate speech frames during call setup or call handover.
  • the device for controlling the voice coding rate, the base station control device, and the base transceiver station provided by the foregoing embodiment are only illustrated by the division of the foregoing functional modules when controlling the voice coding rate.
  • the above function assignment is completed by different functional modules, that is, the internal structure of the device is divided into different functional modules to complete all or part of the functions described above.
  • the setting of the control speech coding rate provided by the foregoing embodiment The system, the base station control device, the base transceiver station, the system for controlling the voice coding rate, and the method for controlling the voice coding rate are in the same concept. For details, refer to the method embodiment, and details are not described herein.
  • All or part of the steps in the embodiment of the present invention may be implemented by software, and the corresponding software program may be stored in a readable storage medium such as an optical disk or a hard disk.

Landscapes

  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Computational Linguistics (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Mobile Radio Communication Systems (AREA)
  • Telephone Function (AREA)

Abstract

本发明公开了一种控制语音编码速率的方法、设备及系统，属于通信领域。所述方法包括：接收呼叫预切换通知，所述呼叫预切换通知中携带了本端移动终端MS呼叫切换后采用的语音编码类型；根据所述本端MS呼叫切换后采用的语音编码类型设定调速指示，并将所述调速指示发送给对端MS，使所述对端MS按照所述调速指示进行语音编码，直至所述本端MS完成呼叫切换。所述设备包括：接收模块、设定模块和发送模块。本发明通过根据本端MS呼叫切换后采用的语音编码类型设定调速指示，并将调速指示发送给对端MS，使对端MS按照该调速指示进行语音编码，避免了本端MS因无法接听对端MS发送的高速率语音帧而导致的静音问题。

Description

控制语音编码速率的方法、 设备及系统 本申请要求于 2010年 11月 02 日提交中国专利局、 申请号为
201010532385. 8 , 发明名称为 "控制语音编码速率的方法、 设备及系统 " 的中国专利申请的优先权， 全部内容通过引用结合在本申请中。 技术领域
本发明涉及通信领域， 特别涉及一种控制语音编码速率的方法、 设备 及系统。 背景技术
在 GSM ( Globa l Sys tem for Mob i le communicat ions , 全球移动通讯 系统） 中引入 AoIP ( A接口用户面承载 IP化）之后， 为了提升语音质量， 一 个重大的改进是参与呼叫的双方 MS ( Mobi le Stat ion , 移动终端）采用兼 容语音编码， 使语音数据无需经 TC ( Transcoder , 编解码器）做任何处理， 即可实现端到端的语音数据透传， 即实现 TrFO ( Transcoder Free Operat ion, 免编解码操作） 。 由此可见， 在 AoIP中实现 TrFO的前提是呼叫 双方采用兼容语音编码。
针对采用兼容的 AMR ( Adapt ive Mul t i-Rate , 自适应多速率） 全半速 率集实现 TrFO的场景， 由于一个 AMR速率集中包含多个速率， 现有技术中参 与呼叫的 MS根据空口质量确定采用的语音编码速率。 当空口质量比较好的 时候， 可以选用高的语音编码速率； 当空口质量比较差的时候， 可以选用 较低的语音编码速率。
发明人发现现有技术至少存在以下缺点：
由于现有技术中参与呼叫的 MS是根据空口质量来确定采用的语音编码 速率， 因而在某些场景下， 可能由于呼叫一端的空口质量好而采用了超出 对端所允许的最高速率， 导致高速率的语音帧无法下发给对端， 从而造成 静音。 例如， 在呼叫切换场景， 呼叫双方 MS1和 MS2在发生呼叫切换前均采 用全速率集维持 TrFO呼叫， 当 MS1发生呼叫切换后采用兼容的半速率集编码 维持 TrFO呼叫时， 如果 MS2采用了全速集中的最高编码速率进行语音编码， 贝 在呼叫切换时将无法接收到 MS2发送的高速语音帧， 导致静音。 发明内容
为了避免因高速率语音帧无法下发而导致的静音问题， 本发明实施例 提供了一种控制语音编码速率的方法、 设备及系统。 所述技术方案如下： 一方面， 提供了一种控制语音编码速率的方法， 所述方法包括： 接收呼叫预切换通知， 所述呼叫预切换通知中携带了本端移动终端 MS 呼叫切换后采用的语音编码类型；
根据所述本端 MS呼叫切换后采用的语音编码类型设定调速指示， 并将 所述调速指示发送给对端 MS ,使所述对端 MS按照所述调速指示进行语音编 码， 直至所述本端 MS完成呼叫切换。
另一方面， 提供了一种控制语音编码速率的设备， 所述设备包括： 接收模块， 用于接收呼叫预切换通知， 所述呼叫预切换通知中携带了 本端移动终端 MS呼叫切换后采用的语音编码类型；
设定模块， 用于根据所述接收模块接收到的本端 MS呼叫切换后采用的 语音编码类型设定调速指示；
发送模块， 用于将所述设定模块设定的调速指示发送给对端 MS , 使所 述对端 MS按照所述调速指示进行语音编码， 直至所述本端 MS完成呼叫切 换。
还提供了一种控制语音编码速率的方法， 所述方法包括：
接收语音帧；
将所述语音帧中携带的调速指示修改为默认初始速率， 并将携带了所 述默认初始速率的语音帧发送给对应的移动终端 MS ,使所述对应的 MS根据 所述默认初始速率进行语音编码。
还提供了一种基站控制设备， 所述基站控制设备包括：
接收模块， 用于接收语音帧；
修改模块， 用于将所述接收模块接收到的语音帧中携带的调速指示修 改为默认初始速率；
发送模块， 用于将所述修改模块修改后的携带了默认初始速率的语音 帧发送给对应的移动终端 MS ,使所述对应的 MS根据所述默认初始速率进行 语音编码。
还提供了一种基站收发台， 所述基站收发台包括： 接收模块， 用于接收语音帧；
修改模块， 用于将所述接收模块接收到的语音帧中携带的调速指示修 改为默认初始速率；
发送模块， 用于将所述修改模块修改后的携带了默认初始速率的语音 帧发送给对应的移动终端 MS ,使所述对应的 MS根据所述默认初始速率进行 语音编码。
还提供了一种控制语音编码速率的系统， 所述系统包括： 基站控制设 备和基站收发台；
所述基站控制设备如上述基站控制设备； 所述基站收发台如上述基站 收发台。
本发明实施例提供的技术方案的有益效果是：
通过根据本端 MS呼叫切换后采用的语音编码类型设定调速指示， 并将 调速指示发送给对端 MS ,使对端 MS按照该调速指示进行语音编码，从而避 免了本端 MS在呼叫切换时的短时间内因无法接听对端 MS发送的高速率语 音帧而导致的静音问题； 另外， 通过将接收到的语音帧中携带的调速指示 修改为默认初始速率， 并将其发送给对应的 MS ,使对应的 MS按照默认初始 速率进行语音编码， 从而进一步避免了因无法接听高速率语音帧而导致的 静音问题。 附图说明
为了更清楚地说明本发明实施例中的技术方案， 下面将对实施例描述 中所需要使用的附图作筒单地介绍， 显而易见地， 下面描述中的附图仅仅 是本发明的一些实施例， 对于本领域普通技术人员来讲， 在不付出创造性 劳动的前提下， 还可以根据这些附图获得其他的附图。
图 1是本发明实施例一提供的控制语音编码速率的方法流程图； 图 2是本发明实施例二提供的控制语音编码速率的方法流程图； 图 3是本发明实施例三提供的呼叫切换场景示意图；
图 4是本发明实施例三提供的呼叫切换场景消息交互示意图； 图 5是本发明实施例三提供的控制语音编码速率的方法流程图； 图 6是本发明实施例三提供的另一种呼叫切换场景示意图；
图 7是本发明实施例三提供的另一种呼叫切换场景消息交互示意图； 图 8是本发明实施例三提供的另一种控制语音编码速率的方法流程图； 图 9是本发明实施例三提供的呼叫建立场景示意图；
图 10 是本发明实施例三提供的又一种控制语音编码速率的方法流程 图；
图 11是本发明实施例四提供的控制语音编码速率的设备结构示意图； 图 12是本发明实施例四提供的发送模块结构示意图；
图 1 3是本发明实施例四提供的另一种控制语音编码速率的设备结构示 意图；
图 14是本发明实施例五提供的基站控制设备结构示意图；
图 15是本发明实施例六提供的基站收发台结构示意图；
图 16是本发明实施例六提供的另一种基站收发台结构示意图； 图 17是本发明实施例七提供的控制语音编码速率的系统结构示意图。 具体实施方式
为使本发明的目的、 技术方案和优点更加清楚， 下面将结合附图对本 发明实施方式作进一步地详细描述。
实施例一
参见图 1 , 本实施例提供了一种控制语音编码速率的方法， 该方法流程 具体如下：
101 : 接收呼叫预切换通知， 该呼叫预切换通知中携带了本端 MS呼叫 切换后采用的语音编码类型；
102 : 根据本端 MS呼叫切换后采用的语音编码类型设定调速指示； 103: 将该调速指示发送给对端 MS , 使对端 MS按照该调速指示进行语 音编码， 直至本端 MS完成呼叫切换。
具体地， BTS ( Ba se Trans ce iver S ta t i on , 基站收发台 )根据本端 MS 呼叫切换后的语音编码类型设定调速指示， 具体包括：
BTS根据本端 MS呼叫切换后采用的语音编码类型确定呼叫切换后的默 认初始速率， 并将呼叫切换后的默认初始速率作为调速指示。
其中， 本端 MS呼叫切换后采用的语音编码类型给出了本端 MS切换后 采用的 AMR速率集， 对于包含多个速率的 AMR速率集， 协议作出了如下选 择初始呼叫速率的规定： 如果速率集包含 4个速率， 则初始呼叫速率为次低速率； 如果速率集包含 2个或 3个速率， 则初始呼叫速率为最低速率。
本实施例可以按照上述协议规定的方式， 将上述初始呼叫速率作为呼 叫切换后的默认初始速率。 当然， 还可以选择其它的确定方式， 本实施例 不对确定呼叫切换后的默认初始速率的方式及确定的具体值进行限定， 能 够保证本端 MS在呼叫切换后仍能正常接听到对端 MS发送的语音帧即可。
进一步地， 将该调速指示发送给对端 MS , 具体包括：
BTS接收本端 MS在呼叫切换前发送给对端 MS的语音帧，语音帧中携带 了本端 MS根据本端空口质量确定的调速指示；
将语音帧中携带的调速指示修改为根据本端 MS呼叫切换后采用的语音 编码类型设定的调速指示， 并将修改后的语音帧发送给对端控制设备， 由 对端控制设备将修改后的语音帧发送给对端 MS。
对于本端 MS在呼叫切换前未向对端 MS发送语音帧的静音场景， 将该 调速指示发送给对端 MS , 具体包括：
如果在预设时间内 BTS未接收到本端 MS在呼叫切换前发送给对端 MS 的语音帧， 则向对端控制设备 BTS发送携带了该调速指示的静音帧， 由对 端 BTS将静音帧修改为填充帧后发送给对端 MS。
其中， 静音帧具体可以为 No_Data帧， 填充帧具体可以为 SID_Fi l ler 帧， 本实施例及后续实施例涉及到的预设时间可根据实际情况进行设定， 例如， 将预设时间设定为 20毫秒， 本实施例不对预设时间的具体值进行限 定。 也就是说， 如果在预设时间内未接收到本端 MS在呼叫切换前发送给对 端 MS的语音帧， 则向对端控制设备发送携带了该调速指示的 No_Data帧， 由对端控制设备将接收到的携带了该调速指示的 No_Da ta 帧修改为
SID-Fi l ler帧后发送给对端 MS , 使对端 MS按照 SID_Fi l ler帧中携带的该 调速指示进行语音编码， 从而实现速率调整。
可选地， 根据本端 MS呼叫切换后采用的语音编码类型设定调速指示之 前， 还包括：
判断本端 MS呼叫切换后采用的接口承载类型是否为 IP承载； 如果是， 则执行根据本端 MS呼叫切换后采用的语音编码类型设定调速 指示的步骤。
需要说明的是， 本实施例及后续实施例均以本端 MS发起呼叫建立或进 行呼叫切换为例， 对控制语音编码速率的方法进行了说明。 实际应用中， 还可以由对端 MS发起呼叫建立或进行呼叫切换， 其实现过程与本端 MS的 实现过程相同， 此处不再赘述。
本实施例提供的方法， 通过根据本端 MS呼叫切换后采用的语音编码类 型设定调速指示， 并将调速指示发送给对端 MS ,使对端 MS按照该调速指示 进行语音编码， 或在对端 MS接听呼叫前， 通过将核心网下发的语音帧中的 调速指示修改为本端默认初始速率， 并发送给本端 MS ,使本端 MS根据本端 默认初始速率进行语音编码， 从而避免了本端 MS无法接听对端 MS发送的 高速率语音帧或对端 MS无法接收高速率语音帧而导致的静音问题， 为双方 能够进行正常呼叫提供了保证。 实施例二
本实施例提供了一种控制语音编码速率的方法， 参见图 2 , 该方法流程 具体如下：
201 : 接收语音帧；
202: 将语音帧中携带的调速指示修改为默认初始速率；
203: 将携带了默认初始速率的语音帧发送给对应的 MS , 使对应的 MS 根据默认初始速率进行语音编码。
其中， 针对呼叫建立场景， 接收语音帧， 具体包括：
BCS ( Base S ta t ion Cont ro l ler , 基站控制器）接收核心网在放回铃 音过程中下发的语音帧， 语音帧中携带了由核心网根据主叫 MS空口质量确 定的调速指示；
相应地， 将语音帧中携带的调速指示修改为默认初始速率， 并将携带 了默认初始速率的语音帧发送给对应的移动终端 MS , 具体包括：
BCS将核心网下发的语音帧中携带的调速指示修改为主叫 MS的默认初 始速率， 并将携带了主叫 MS的默认初始速率的语音帧发送给主叫 MS , 使主 叫 MS按照主叫 MS的默认初始速率进行语音编码。
针对呼叫切换场景， 接收语音帧， 具体包括：
基站收发台 BTS接收本端 MS在呼叫切换前发送给对端 MS的语音帧， 语音帧中携带了本端 MS根据本端空口质量确定的调速指示；
相应地， 将语音帧中携带的调速指示修改为默认初始速率， 并将携带 了默认初始速率的语音帧发送给对应的移动终端 MS , 具体包括：
BTS将本端 MS发送的语音帧中携带的调速指示修改为本端 MS呼叫切换 后的默认初始速率， 并将携带了本端 MS呼叫切换后的默认初始速率的语音 帧发送给对端 MS , 使对端 MS按照本端 MS呼叫切换后的默认初始速率进行 语音编码。
进一步地， BTS将本端 MS发送的语音帧中携带的调速指示修改为本端 MS呼叫切换后的默认初始速率之前， 还包括：
BTS接收基站控制设备发送的呼叫预切换通知，该呼叫预切换通知中携 带了本端 MS呼叫切换后采用的语音编码类型；
BTS根据本端 MS呼叫切换后采用的语音编码类型确定本端 MS呼叫切换 后的默认初始速率；
相应地， BTS将本端 MS发送的语音帧中携带的调速指示修改为本端 MS 呼叫切换后的默认初始速率， 具体包括：
将本端 MS发送的语音帧中携带的调速指示修改为根据本端 MS呼叫切 换后采用的语音编码类型确定的本端 MS呼叫切换后的默认初始速率。
本实施例提供的方法， 通过将接收到的语音帧中携带的调速指示修改 为默认初始速率， 并将其发送给对应的 MS ,使对应的 MS按照默认初始速率 进行语音编码， 从而避免了在呼叫建立或呼叫切换过程中因 MS无法接听高 速率语音帧而导致的静音问题。 实施例三
本实施例提供了一种控制语音编码速率的方法， 为了便于说明， 本实 施例首先以图 3所示的 BSC内呼叫切换场景为例， 结合图 4所示的消息交 互过程， 对本实施例提供的控制语音编码速率的方法进行详细描述。
图 3中， 呼叫双方 MS1和 MS2均在相同的 BSC下， 切换前 MS1和 MS2 都采用全速率集 AMR—FR Se t l ( 12. 2kpbs、 7. 4kpbs、 5. 9kpbs、 4. 95kpbs ) 建立 TrFO呼叫。 后续 MSI发生 BSC内呼叫切换后， 将采用兼容的半速率集 AMR-HR Se t l ( 7. 4kpbs、 5. 9kpbs、 4. 95kpbs )编码， 维持 TrFO呼叫。 如 果呼叫切换时， MS2采用了 12. 2kpbs的编码速率， 对于呼叫切换后最高语 音编码速率仅为 7. 40kpbs的 MSI来讲，将导致切换到新信道后无法接收 MS2 发送的高速语音帧， 造成切换后的短时间静音。 静音状态直至持续到 MS1 切换完成后， 通过 MSI向 MS2发送调速指示 CMR (Code Mode Reques t , 速率 调整请求）通知 MS2降低编码速率， MS2将语音速率低于 12. 2kbps之后， MSI 方能接收到来自 MS2的语音帧。
针对该种情况，为了避免 MS1在进行 BSC内切换时由于无法接收到 MS2 发送的高速语音帧而出现静音， 本实施例提供了一种控制语音编码速率的 方法， 参见图 5 , 本实施例提供的方法流程具体如下：
501 : BTS1接收 BSC发送的呼叫预切换通知， 该呼叫预切换通知中携带 了 MS1呼叫切换后采用的语音编码类型；
针对该步骤，结合图 4 ,在 BSC接收到 MS1发送的测量报告 Measurement Repor t后， 由于是 BSC内发生呼叫切换， BSC能够确定 MS1在呼叫切换前 后采用的语音编码类型。对于本实施例， MS1呼叫切换前后采用的语音编码 类型从全速率切换到半速率， 则 BSC确定 MS1呼叫切换前后发生了 AMR全 半速率切换， 且 MS1 呼叫切换前后采用的语音编码类型兼容， 则通过呼叫 预切换通知 Pre-Handover Not if icat ion通知 BTSl该 MSI将发生呼叫切换， 并为了 BTS1能够确定 MS1呼叫切换后采用的语音编码速率， BSC将 MS1呼 叫切换后采用的语音编码类型携带在 Pre-Handover Not if icat ion 中发送 给 BTS1 , 之后再向 MS1 下发切换命令 Handover Command/分配命令
As s ignment Command。
502: BTSl根据 MSI呼叫切换后采用的语音编码类型设定调速指示； 具体地， 本实施例不对 BTS1根据 MS1呼叫切换后采用的语音编码类型 设定调速指示的方式进行具体限定。 按照实施例一中描述的调速指示的设 定方式， 由于该 MS1 呼叫切换后采用的语音编码类型为 AMR_HR Set l ( 7. 4kpbs、 5. 9kpbs、 4. 95kpbs ), 则 BTSl设定的调速指示应为 3个速率 中的最低速率 4. 95kpbs。
503: BTSl接收 MSI在呼叫切换前发送给 MS2的语音帧， 该语音帧中携 带了 MS1根据本端空口质量确定的调速指示；
其中， 本实施例不限定该步骤 503与上述步骤 502的先后顺序， 实际 应用中， BTS1还可以先接收 MS1在呼叫切换前发送给 MS2的语音帧， 再根 据 MS1呼叫切换后采用的语音编码类型设定调速指示。
504: BTS1将接收到的语音帧中携带的调速指示修改为根据 MS1呼叫切 换后采用的语音编码类型设定的调速指示，并将修改后的语音帧发送给 MS2 对应的 BTS2 ;
505 : BTS2将该修改后的语音帧发送给 MS2 , 使 MS2按照 BTS1设定的 调速指示进行语音编码。
对于步骤 504和步骤 505 , 由于 BTS1在步骤 504中将 MS1在呼叫切换 前发送给 MS2的语音帧中的调速指示修改成了 BTS1在上述步骤 502中设定 的调速指示，而该调速指示为呼叫切换后的默认初始速率，则通过步骤 505 , BTS2将该修改后的语音帧发送给 MS2 , 使 MS2按照 BTS1设定的调速指示进 行语音编码， 即使 MS1 呼叫切换后采用半速率， 仍然能够正常接收到 MS2 根据呼叫切换后的默认初始速率进行编码的语音帧， 从而避免了 MS1 在呼 叫切换时发生短时间静音， 还能够提高抗干扰能力。 但由于呼叫切换后的 默认初始速率对于 MS2采用的最高编码速率 12. 2kpbs而言相对较低， 则当 MS1完成呼叫切换后， 还可以通过 MS1和 MS2之间的自主调速， 将编码速率 由呼叫切换后的默认初始速率更改为更高的编码速率， 从而提高语音质量。 例如， 更改为 MS1呼叫切换后的最高编码速率 7. 4kpbs。 接下来， 为了更全面的介绍本实施例提供的方法， 本实施例将以图 6 所示的 BSC 间呼叫切换场景为例， 对本实施例提供的控制语音编码速率的 方法做进一步详细描述。
图 6中， 呼叫双方 MS1和 MS2均在相同的 BSC1下， MS1和 MS2都采用 AMR-FR Se t l , 建立 TrFO呼叫。 后续 MS1发生 BSC间切换 (从 BSC1切换到 BSC )后， 将采用兼容的 AMR_HR Set l编码， 维持 TrFO呼叫。 如果呼叫 切换时， MS2采用了 12. 2kpbs的编码速率， 对于呼叫切换后最高语音编码 速率仅为 7. 40kpbs的 MS1来讲， 将导致切换到新信道后无法接收 MS2发送 的高速语音帧， 造成切换后的短时间静音。 静音状态直至持续到 MS1 切换 完成后， 通过 CMR通知 MS2降低编码速率， MS2将语音速率低于 12. 2kbps 之后， MS1方能接收到来自 MS2的语音帧。
针对该种情况，为了避免 MS1在进行 BSC间切换时由于无法接收到 MS2 发送的高速语音帧而出现静音， 结合图 7 所示的消息交互过程， 本实施例 提供的方法通过对 BSC间呼叫切换的消息进行扩展， 使 MS1呼叫切换后对 应的 BSC1'将 MS1呼叫切换后采用的语音编码类型通知给 BSC1 , 从而实现 对语音编码速率的控制。 参见图 8 , 本实施例提供的控制语音编码速率的方 法流程具体如下：
801: MSC (Mobile Switching Center, 移动交换中心）接收 MSI呼叫 切换后对应的 BSC1'发送的切换请求确认，该切换请求确认中携带了 MS1呼 叫切换后采用的语音编码类型；
具体地， 由于是 BSC间发生呼叫切换， 发生呼叫切换的 MS1所对应的
BSC1预先并不确定 MS1在呼叫切换后采用何种语音编码类型， 因此， 本实 施例提供的方法在具体实现时，通过 BSC1'将 MS1呼叫切换后采用的语音编 码类型携带在切换请求确认 HANDOVER REQUEST ACKN0WLEGE 中发送给对应 的 MSC, 使 MSC在收到 BSC1'发送的 HANDOVER REQUEST ACKN0WLEGE之后， 得知 MSI呼叫切换之后将采用的语音编码类型 Codec。
802: MSC将 MSI呼叫切换后采用的语音编码类型携带在切换命令中一 并发送给 BSC 1;
针对该步骤， MSC 向 BSC1 发送切换命令 HANDOVER COMMAND, 通过在 HANDOVER COMMAND中加入 MSI呼叫切换后采用的编码类型指示信息，使 BSC1 能够获知 MS1 呼叫切换后采用的语音编码类型。 其中， 本实施例不对 HANDOVER COMMAND中加入 MSI呼叫切换后采用的编码类型指示信息的方式 进行限定。 例如， 可以在现有 A口 HANDOVER C0匪 AND消息中， 扩展 NewBSS to Old BSS Information信元， 力口入 Speech Codec ( Chosen )字段， 用以 指示 MSI呼叫切换后将采用的编解码类型。
803: BSC1接收到 MSC发送的切换命令后， 向 BTS1发送呼叫预切换通 知， 该呼叫预切换通知中携带了 MS1呼叫切换后采用的语音编码类型； 其中， BSC1接收到 MSC发送的 HANDOVER COMMAND后， 如果 MS1呼叫切 换前采用了全速率 AMR编码， 则 BSC1可以得知 MS1呼叫切换后采用与切换 前 AMR速率集兼容的半速率信道。 BSC1通过将 MS1呼叫切换后采用的语音 编码类型携带在 Pre- Handover Notification中发送给 BTS1, 触发 BTS1在 向对端 MS2发送的语音帧时， 将语音帧中的调速指示设定为呼叫切换后的 默认初始速率， 之后 BCS1 再向 MS1 下发 Handover Command/Assignment Command。
可选地， BSC1'在向 MSC发送的 HANDOVER REQUEST ACKN0WLEGE中， 还 可以携带接口承载类型， 则 BSC1需要对 MS1呼叫切换前后的接口承载类型 是否为 IP 进行判断， 在判断切换后采用 IP 承载后， 再向 BTS1 下发 Pre- Handover Notification 当然， BSC1也可以省略判断步骤， 本实施例 对此不作具体限定。
804: BTS1根据呼叫预切换通知中携带的 MS1呼叫切换后采用的语音编 码类型设定调速指示；
805: BTS1接收 MS1在呼叫切换前发送给 MS2的语音帧， 该语音帧中携 带了 MS1根据本端空口质量确定的调速指示；
806: BTS1将接收到的语音帧中携带的调速指示修改为根据 MS1呼叫切 换后采用的语音编码类型设定的调速指示，并将修改后的语音帧发送给 MS2 对应的 BTS2;
807: BTS2将该修改后的语音帧发送给 MS2, 使 MS2按照 BTS1设定的 调速指示进行语音编码。
具体地， 步骤 804至步骤 807同步骤 502至步骤 505, 本实施例在此不 再赘述。 需要说明的是， 图 8所示的 BSC间呼叫切换是在同一 MSC下进行 的呼叫切换， 对于不同 MSC之间的呼叫切换场景， 本实施例提供的方法同 样支持， 实现方式与图 8所示的 BSC间呼叫切换原理相同， 此处同样不再 赘述。
可选地， 无论是 BSC内发生呼叫切换， 还是 BSC间发生呼叫切换， 如果 即将发生呼叫切换的 MS1在呼叫切换前没有向 MS2发送语音帧， 即在 DTX ( Discontinuous transmission, 断续传输）打开的场景下， 此时 BTS1在 根据 MS1呼叫切换后采用的语音编码类型设定调速指示后， BTS1将采用区别 于上述步骤 504或步骤 804的方式， 将设定的调速指示发送给 MS2。 具体实现 时， BTS1将其设定的调速指示携带在 No_Data帧中发送给 MS2对应的 BTS2, BTS2在将该 No_data帧转化为 SID_Filler帧后发送给 MS 2, 使 MS 2同样能够根 据 BTS1设定的调速指示进行语音编码， 从而达到速率调整的目的。
另外， 无论呼叫双方是位于同一 BSC内， 或是位于同一 MSC内的不同 BSC 下，还是位于不同 MSC间，呼叫双方在建立呼叫时，如果被叫端未接听呼叫， 都将触发核心网向主叫端放回铃音。 以图 9所示的呼叫建立场景为例， 在被 叫 MS2接听呼叫之前， 核心网向主叫 MS1放回铃音， 因为空口质量比较好， MS1采用了最高的 12.2kbps的语音编码速率。 MS2接听后， 来自 MS1的 12.2kbps的语音帧无法通过半速率空口信道下发给 MS2, 导致一些高速率的 语音帧丟失。 直到被叫用户接听， 来自 MS2的语音帧携带调速指示， 要求主 叫 MS I调整速率至 12. 2kbps以下， 主叫用户的语音帧才能被被叫接收。
针对该种情况， 为了避免在进行呼叫建立时 MS2因无法接收到 MS 1发送 的高速语音帧而出现静音， 参见图 1 0 , 本实施例提供的方法还包括：
1 001 : 在被叫 MS接听呼叫之前， 主叫侧 BSC接收核心网在放回铃音过 程中下发的语音帧， 该语音帧中携带了由核心网根据主叫侧空口质量确定 的调速指示；
针对该步骤， 即使主叫侧空口质量好， 使核心网下发的语音帧中携带 的调速指示为较高的语音编码速率， 但通过后续步骤由主叫侧 BSC将核心 网下发的语音帧中的调速指示进行修改后，被叫 MS同样能够接收到主叫 MS 发送的语音帧。
1 002 : 主叫侧 BSC将核心网下发的语音帧中携带的调速指示修改为主 叫侧默认初始速率；
具体地， 该步骤中的主叫侧默认初始速率的确定方式同样可采用上述 一中提供的默认初始速率的确定方式， 本实施例对此不作具体限定。
Figure imgf000014_0001
例如， 图 9中， 作为被叫的 MS2接听呼叫之前， 主叫侧 BSC1接收核心 网即 MSC 1在放回铃音过程中下发的语音帧， 该语音帧中携带了 MSC 1根据 主叫侧空口质量确定的调速指示； BSC1将接收到的 MSC1下发的语音帧中携 带的调速指示修改为本端默认初始速率， 并发送给 BTS 1 ,再由 BTS 1发送给 MS 1，使 MS 1根据本端默认初始速率进行语音编码。 由于 MS 1所在主叫侧的 默认初始速率不高于 MS 2采用的编码速率， 则 MS 2将能够成功接收到 MS 1 发送的语音帧， 从而避免了呼叫建立过程中出现静音。
需要说明的是， 由于核心网下发的语音帧中携带的调速指示是根据主 叫侧空口质量确定的， 随着主叫侧空口质量的变化， 核心网下发的语音帧 中携带的调速指示是可变的。 除此之外， 核心网还可以选择在语音帧中携 带固定的调速指示， 例如， 核心网将下发的语音帧中携带的调速指示设定 为主叫侧默认初始速率， 针对此种情况， 主叫侧在判断出核心网下发的语 音帧中携带的调速指示为本端默认初始速率之后， 无需对核心网下发的语 音帧进行修改， 同样能够达到主叫 MS按照本端默认初始速率进行语音编码 的目的。 关于核心网下发的语音帧中具体携带何种调速指示， 本实施例对 此不作限定。
本实施例提供的方法， 通过根据本端 MS呼叫切换后采用的语音编码类 型设定调速指示， 并将调速指示发送给对端 MS ,使对端 MS按照该调速指示 进行语音编码， 或在对端 MS接听呼叫前， 通过将核心网下发的语音帧中的 调速指示修改为本端默认初始速率， 并发送给本端 MS ,使本端 MS根据本端 默认初始速率进行语音编码， 从而避免了本端 MS无法接听对端 MS发送的 高速率语音帧或对端 MS无法接收高速率语音帧而导致的静音问题， 为双方 能够进行正常呼叫提供了保证。 实施例四
参见图 11 , 本实施例提供了一种控制语音编码速率的设备， 该设备用 于执行上述实施例一及实施例三中 BTS设备所执行的方法步骤， 该设备包 括：
接收模块 1101 , 用于接收呼叫预切换通知， 呼叫预切换通知中携带了 本端移动终端 MS呼叫切换后采用的语音编码类型；
设定模块 1102 , 用于根据接收模块 1101接收到的本端 MS呼叫切换后 采用的语音编码类型设定调速指示；
发送模块 1103 , 用于将设定模块 1102设定的调速指示发送给对端 MS , 使对端 MS按照调速指示进行语音编码， 直至本端 MS完成呼叫切换。
其中，设定模块 1102 , 具体用于根据本端 MS呼叫切换后采用的语音编 码类型确定呼叫切换后的默认初始速率， 并将呼叫切换后的默认初始速率 作为调速指示。 关于呼叫切换后的默认初始速率如何确定， 可参见上述实 施例一及上述实施例三中步骤 502的相关描述， 此处不再赘述。
参见图 12 , 发送模块 11 03 , 具体包括：
接收单元 1103a ,用于接收本端 MS在呼叫切换前发送给对端 MS的语音 帧， 语音帧中携带了本端 MS根据本端空口质量确定的调速指示；
修改单元 1103b ,用于将接收单元 1103a接收到的语音帧中携带的调速 指示修改为根据本端 MS呼叫切换后采用的语音编码类型设定的调速指示； 发送单元 1103c ,用于将修改单元 1103b修改后的语音帧发送给对端控 制设备， 由对端控制设备将修改后的语音帧发送给对端 MS。
可选地， 对于静音场景， 发送模块 1103 , 具体用于如果在预设时间内 未接收到本端 MS在呼叫切换前发送给对端 MS的语音帧， 则向对端控制设 备发送携带了调速指示的静音帧， 由对端控制设备将静音帧修改为填充帧 后发送给对端 MS。
静音帧具体可以为 No_Data帧， 填充帧具体可以为 S ID_Fi l ler帧， 本 实施例及后续实施例涉及到的预设时间可根据实际情况进行设定， 例如， 将预设时间设定为 20毫秒， 本实施例不对预设时间的具体值进行限定。 也 就是说， 如果在预设时间内未接收到本端 MS在呼叫切换前发送给对端 MS 的语音帧， 则向对端控制设备发送携带了该调速指示的 No_Data 帧， 由对 端控制设备将接收到的携带了该调速指示的 No_Data帧修改为 SID_Fi l ler 帧后发送给对端 MS , 使对端 MS按照 SID_Fi l ler帧中携带的该调速指示进 行语音编码， 从而实现速率调整。
可选地， 参见图 13 , 该设备还包括：
判断模块 1104 ,用于判断本端 MS呼叫切换前后采用的接口承载类型是 否为 IP承载；
设定模块 1102 , 用于在判断模块 1104判断出本端 MS呼叫切换后采用 的接口承载为 IP承载后， 执行根据本端 MS呼叫切换后采用的语音编码类 型设定调速指示的步骤。
本实施例提供的设备， 通过根据本端 MS呼叫切换后采用的语音编码类 型设定调速指示， 并将调速指示发送给对端 MS ,使对端 MS按照该调速指示 进行语音编码， 或在对端 MS接听呼叫前， 通过将核心网下发的语音帧中的 调速指示修改为本端默认初始速率， 并发送给本端 MS ,使本端 MS根据本端 默认初始速率进行语音编码， 从而避免了本端 MS无法接听对端 MS发送的 高速率语音帧或对端 MS无法接收高速率语音帧而导致的静音问题， 为双方 能够进行正常呼叫提供了保证。 实施例五
本实施例提供了一种基站控制设备， 该设备用于执行上述实施例二及 实施例三中 BSC所执行的方法步骤， 参见图 14 , 该设备包括：
接收模块 1401 , 用于接收语音帧；
修改模块 1402 ,用于将接收模块 1401接收到的语音帧中携带的调速指 示修改为默认初始速率； 发送模块 1403 ,用于将修改模块 1402修改后的携带了默认初始速率的 语音帧发送给对应的移动终端 MS ,使对应的 MS根据默认初始速率进行语音 编码。
其中， 接收模块 1401 , 具体用于接收核心网在放回铃音过程中下发的 语音帧， 语音帧中携带了由核心网根据主叫 MS空口质量确定的调速指示； 相应地，修改模块 1402 , 具体用于将接收模块 1401接收到的核心网下 发的语音帧中携带的调速指示修改为主叫 MS的默认初始速率；
发送模块 1403 , 具体用于将修改模块 1402修改后的携带了主叫 MS的 默认初始速率的语音帧发送给主叫 MS , 使主叫 MS按照主叫 MS的默认初始 速率进行语音编码。
本实施例提供的基站控制设备， 通过将接收到的核心网下发的语音帧 中携带的调速指示修改为主叫 MS的默认初始速率， 并将其发送给主叫 MS , 使主叫 MS按照默认初始速率进行语音编码， 从而避免了在呼叫建立过程中 因被叫 MS无法接听主叫 MS发送的高速率语音帧而导致的静音问题。 实施例六
本实施例提供了一种基站收发台， 该基站收发台用于执行上述实施例 二及实施例三中 BTS所执行的方法步骤， 参见图 15 , 该基站收发台包括： 第一接收模块 1501 , 用于接收语音帧；
修改模块 1502 ,用于将第一接收模块 1501接收到的语音帧中携带的调 速指示修改为默认初始速率；
发送模块 1503 ,用于将修改模块 1502修改后的携带了默认初始速率的 语音帧发送给对应的移动终端 MS ,使对应的 MS根据默认初始速率进行语音 编码。
其中， 第一接收模块 1501 , 具体用于接收本端 MS在呼叫切换前发送给 对端 MS的语音帧， 语音帧中携带了本端 MS根据本端空口质量确定的调速 指示；
相应地，修改模块 1502 , 具体用于将第一接收模块 1501接收到的本端 MS发送的语音帧中携带的调速指示修改为本端 MS呼叫切换后的默认初始速 率；
发送模块 1503 , 具体用于将修改模块 1502修改后的携带了本端 MS呼 叫切换后的默认初始速率的语音帧发送给对端 MS , 使对端 MS按照本端 MS 呼叫切换后的默认初始速率进行语音编码。
进一步地， 参见图 16 , 该基站收发台还包括：
第二接收模块 1504 , 用于接收基站控制设备发送的呼叫预切换通知， 该呼叫预切换通知中携带了本端 MS呼叫切换后采用的语音编码类型；
确定模块 1505 , 用于根据第二接收模块 1504接收到的本端 MS呼叫切 换后采用的语音编码类型确定本端 MS呼叫切换后的默认初始速率；
相应地，修改模块 1502 , 具体用于将第一接收模块 1501接收到的本端 MS发送的语音帧中携带的调速指示修改为确定模块 1505确定的本端 MS呼 叫切换后的默认初始速率。
本实施例提供的基站收发台， 通过在接收到基站控制设备发送的呼叫 预切换通知后， 将接收到的本端 MS发送的语音帧中携带的调速指示修改为 本端 MS呼叫切换后的默认初始速率， 并将其发送给对端 MS , 使对端 MS按 照本端 MS呼叫切换后的默认初始速率进行语音编码， 从而避免了在呼叫切 换过程中因本端 MS无法接听对端 MS发送的高速率语音帧而导致的静音问 题。 实施例七
本实施例提供了一种控制语音编码速率的系统， 参见图 17 , 该系统包 括： 基站控制设备 1701和基站收发台 1702 ;
其中， 基站控制设备 1701如上述实施例五提供的基站控制设备， 基站 收发台 1702如上述实施例六提供的基站收发台。
本实施例提供的系统， 通过基站控制设备或基站收发台将接收到的语 音帧中携带的调速指示修改为默认初始速率， 并将其发送给对应的 MS , 使 对应的 MS按照默认初始速率进行语音编码， 从而避免了在呼叫建立或呼叫 切换过程中因 MS无法接听高速率语音帧而导致的静音问题。
需要说明的是： 上述实施例提供的控制语音编码速率的设备、 基站控 制设备和基站收发台在控制语音编码速率时， 仅以上述各功能模块的划分 进行举例说明， 实际应用中， 可以根据需要而将上述功能分配由不同的功 能模块完成， 即将设备的内部结构划分成不同的功能模块， 以完成以上描 述的全部或者部分功能。 另外， 上述实施例提供的控制语音编码速率的设 备、 基站控制设备、 基站收发台、 控制语音编码速率的系统与控制语音编 码速率的方法实施例属于同一构思， 其具体实现过程详见方法实施例， 这 里不再赘述。
上述本发明实施例序号仅仅为了描述， 不代表实施例的优劣。
本发明实施例中的全部或部分步骤， 可以利用软件实现， 相应的软件 程序可以存储在可读取的存储介质中， 如光盘或硬盘等。
以上所述仅为本发明的较佳实施例， 并不用以限制本发明， 凡在本发 明的精神和原则之内， 所作的任何修改、 等同替换、 改进等， 均应包含在 本发明的保护范围之内。

Claims

权 利 要 求
1、 一种控制语音编码速率的方法， 其特征在于， 所述方法包括： 接收呼叫预切换通知， 所述呼叫预切换通知中携带了本端移动终端 MS 呼叫切换后采用的语音编码类型；
根据所述本端 MS呼叫切换后采用的语音编码类型设定调速指示， 并将 所述调速指示发送给对端 MS ,使所述对端 MS按照所述调速指示进行语音编 码， 直至所述本端 MS完成呼叫切换。
2、 根据权利要求 1 所述的方法， 其特征在于， 所述根据所述本端 MS 呼叫切换后的语音编码类型设定调速指示， 具体包括：
根据所述本端 MS呼叫切换后采用的语音编码类型确定呼叫切换后的默 认初始速率， 并将所述呼叫切换后的默认初始速率作为调速指示。
3、 根据权利要求 1所述的方法， 其特征在于， 所述将所述调速指示发 送给对端 MS , 具体包括：
接收所述本端 MS在呼叫切换前发送给所述对端 MS的语音帧， 所述语 音帧中携带了所述本端 MS根据本端空口质量确定的调速指示；
将所述语音帧中携带的调速指示修改为根据所述本端 MS呼叫切换后采 用的语音编码类型设定的调速指示， 并将修改后的语音帧发送给对端控制 设备， 由所述对端控制设备将所述修改后的语音帧发送给所述对端 MS。
4、 根据权利要求 1所述的方法， 其特征在于， 所述将所述调速指示发 送给对端 MS , 具体包括：
如果在预设时间内未接收到所述本端 MS在呼叫切换前发送给所述对端 MS的语音帧， 则向对端控制设备发送携带了所述调速指示的静音帧， 由所 述对端控制设备将所述静音帧修改为填充帧后发送给所述对端 MS。
5、 根据权利要求 1至 4任一权利要求所述的方法， 其特征在于， 所述 根据所述本端 MS呼叫切换后采用的语音编码类型设定调速指示之前， 还包 括：
判断所述本端 MS呼叫切换后采用的接口承载类型是否为网际协议 IP 承载；
如果是， 则执行根据所述本端 MS呼叫切换后采用的语音编码类型设定 调速指示的步骤。
6、 一种控制语音编码速率的设备， 其特征在于， 所述设备包括： 接收模块， 用于接收呼叫预切换通知， 所述呼叫预切换通知中携带了 本端移动终端 MS呼叫切换后采用的语音编码类型；
设定模块， 用于根据所述接收模块接收到的本端 MS呼叫切换后采用的 语音编码类型设定调速指示；
发送模块， 用于将所述设定模块设定的调速指示发送给对端 MS , 使所 述对端 MS按照所述调速指示进行语音编码， 直至所述本端 MS完成呼叫切 换。
7、 根据权利要求 6所述的设备， 其特征在于， 所述设定模块， 具体用 于根据所述本端 MS呼叫切换后采用的语音编码类型确定呼叫切换后的默认 初始速率， 并将所述呼叫切换后的默认初始速率作为调速指示。
8、 根据权利要求 6所述的设备， 其特征在于， 所述发送模块， 具体包 括：
接收单元， 用于接收所述本端 MS在呼叫切换前发送给所述对端 MS的 语音帧， 所述语音帧中携带了所述本端 MS根据本端空口质量确定的调速指 示；
修改单元， 用于将所述接收单元接收到的语音帧中携带的调速指示修 改为根据所述本端 MS呼叫切换后采用的语音编码类型设定的调速指示； 发送单元， 用于将所述修改单元修改后的语音帧发送给对端控制设备， 由所述对端控制设备将所述修改后的语音帧发送给所述对端 MS。
9、 根据权利要求 6所述的设备， 其特征在于， 所述发送模块， 具体用 于如果在预设时间内未接收到所述本端 MS在呼叫切换前发送给所述对端 MS 的语音帧， 则向对端控制设备发送携带了所述调速指示的静音帧， 由所述 对端控制设备将所述静音帧修改为填充帧后发送给所述对端 MS。
10、 根据权利要求 6至 9任一权利要求所述的设备， 其特征在于， 所 述设备， 还包括：
判断模块， 用于判断所述本端 MS呼叫切换后采用的接口承载类型是否 为网际协议 IP承载；
所述设定模块， 用于在所述判断模块判断出所述本端 MS呼叫切换后采 用的接口承载为 IP承载后， 执行根据所述本端 MS呼叫切换后采用的语音 编码类型设定调速指示的步骤。
11、 一种控制语音编码速率的方法， 其特征在于， 所述方法包括： 接收语音帧；
将所述语音帧中携带的调速指示修改为默认初始速率， 并将携带了所 述默认初始速率的语音帧发送给对应的移动终端 MS ,使所述对应的 MS根据 所述默认初始速率进行语音编码。
12、 根据权利要求 11所述的方法， 其特征在于， 所述接收语音帧， 具 体包括：
基站控制设备 BCS接收核心网在放回铃音过程中下发的语音帧， 所述 语音帧中携带了由所述核心网根据主叫 MS空口质量确定的调速指示；
相应地， 所述将所述语音帧中携带的调速指示修改为默认初始速率， 并将携带了所述默认初始速率的语音帧发送给对应的移动终端 MS , 具体包 括：
所述 BCS将所述核心网下发的语音帧中携带的调速指示修改为主叫 MS 的默认初始速率， 并将携带了所述主叫 MS的默认初始速率的语音帧发送给 所述主叫 MS , 使所述主叫 MS按照所述主叫 MS的默认初始速率进行语音编 码。
13、 根据权利要求 11所述的方法， 其特征在于， 所述接收语音帧， 具 体包括：
基站收发台 BTS接收本端 MS在呼叫切换前发送给对端 MS的语音帧， 所述语音帧中携带了所述本端 MS根据本端空口质量确定的调速指示；
相应地， 所述将所述语音帧中携带的调速指示修改为默认初始速率， 并将携带了所述默认初始速率的语音帧发送给对应的移动终端 MS , 具体包 括：
所述 BTS将所述本端 MS发送的语音帧中携带的调速指示修改为所述本 端 MS呼叫切换后的默认初始速率， 并将携带了所述本端 MS呼叫切换后的 默认初始速率的语音帧发送给对端 MS , 使所述对端 MS按照所述本端 MS呼 叫切换后的默认初始速率进行语音编码。
14、 根据权利要求 13所述的方法， 其特征在于， 所述 BTS将所述本端 MS发送的语音帧中携带的调速指示修改为所述本端 MS呼叫切换后的默认初 始速率之前， 还包括：
所述 BTS接收基站控制设备发送的呼叫预切换通知， 所述呼叫预切换 通知中携带了本端 MS呼叫切换后采用的语音编码类型；
所述 BTS根据所述本端 MS呼叫切换后采用的语音编码类型确定所述本 端 MS呼叫切换后的默认初始速率；
相应地， 所述 BTS将所述本端 MS发送的语音帧中携带的调速指示修改 为所述本端 MS呼叫切换后的默认初始速率， 具体包括：
将所述本端 MS发送的语音帧中携带的调速指示修改为根据所述本端 MS 呼叫切换后采用的语音编码类型确定的所述本端 MS呼叫切换后的默认初始 速率。
15、 一种基站控制设备， 其特征在于， 所述基站控制设备包括： 接收模块， 用于接收语音帧；
修改模块， 用于将所述接收模块接收到的语音帧中携带的调速指示修 改为默认初始速率；
发送模块， 用于将所述修改模块修改后的携带了默认初始速率的语音 帧发送给对应的移动终端 MS ,使所述对应的 MS根据所述默认初始速率进行 语音编码。
16、 根据权利要求 15所述的设备， 其特征在于， 所述接收模块， 具体 用于接收核心网在放回铃音过程中下发的语音帧， 所述语音帧中携带了由 所述核心网根据主叫 MS空口质量确定的调速指示；
相应地， 所述修改模块， 具体用于将所述接收模块接收到的核心网下 发的语音帧中携带的调速指示修改为主叫 MS的默认初始速率；
所述发送模块， 具体用于将所述修改模块修改后的携带了主叫 MS的默 认初始速率的语音帧发送给所述主叫 MS , 使所述主叫 MS按照所述主叫 MS 的默认初始速率进行语音编码。
17、 一种基站收发台， 其特征在于， 所述基站收发台包括：
第一接收模块， 用于接收语音帧；
修改模块， 用于将所述第一接收模块接收到的语音帧中携带的调速指 示修改为默认初始速率；
发送模块， 用于将所述修改模块修改后的携带了默认初始速率的语音 帧发送给对应的移动终端 MS ,使所述对应的 MS根据所述默认初始速率进行 语音编码。
18、 根据权利要求 17所述的基站收发台， 其特征在于， 所述第一接收 模块， 具体用于接收本端 MS在呼叫切换前发送给对端 MS的语音帧， 所述 语音帧中携带了所述本端 MS根据本端空口质量确定的调速指示；
相应地， 所述修改模块， 具体用于将所述第一接收模块接收到的本端 MS发送的语音帧中携带的调速指示修改为所述本端 MS呼叫切换后的默认初 始速率 ^
所述发送模块， 具体用于将所述修改模块修改后的携带了所述本端 MS 呼叫切换后的默认初始速率的语音帧发送给对端 MS ,使所述对端 MS按照所 述本端 MS呼叫切换后的默认初始速率进行语音编码。
19、 根据权利要求 18所述的基站收发台， 其特征在于， 所述基站收发 台还包括：
第二接收模块， 用于接收基站控制设备发送的呼叫预切换通知， 所述 呼叫预切换通知中携带了本端 MS呼叫切换后采用的语音编码类型；
确定模块， 用于根据所述第二接收模块接收到的本端 MS呼叫切换后采 用的语音编码类型确定所述本端 MS呼叫切换后的默认初始速率；
相应地， 所述修改模块， 具体用于将所述第一接收模块接收到的本端
MS发送的语音帧中携带的调速指示修改为所述确定模块确定的本端 MS呼叫 切换后的默认初始速率。
20、 一种控制语音编码速率的系统， 其特征在于， 所述系统包括： 基 站控制设备和基站收发台；
所述基站控制设备如上述权利要求 15或 16所述的基站控制设备； 所 述基站收发台如上述权利要求 17至权利要求 19任一权利要求所述的基站 收发台。
PCT/CN2011/081682 2010-11-02 2011-11-02 控制语音编码速率的方法、设备及系统 WO2012059051A1 (zh)

Priority Applications (1)

Application Number Priority Date Filing Date Title
EP11837577.3A EP2635069A4 (en) 2010-11-02 2011-11-02 METHOD, DEVICE AND SYSTEM FOR CONTROLLING A LANGUAGE CODING RATE

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CN201010532385.8 2010-11-02
CN201010532385.8A CN102143544B (zh) 2010-11-02 2010-11-02 控制语音编码速率的方法、设备及系统

Publications (1)

Publication Number Publication Date
WO2012059051A1 true WO2012059051A1 (zh) 2012-05-10

Family

ID=44410699

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/CN2011/081682 WO2012059051A1 (zh) 2010-11-02 2011-11-02 控制语音编码速率的方法、设备及系统

Country Status (3)

Country Link
EP (1) EP2635069A4 (zh)
CN (1) CN102143544B (zh)
WO (1) WO2012059051A1 (zh)

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2574100A4 (en) * 2010-12-28 2013-08-21 Huawei Tech Co Ltd FREQUENCY ADJUSTMENT METHOD AND DEVICE FOR TRFO LANGUAGE CALL SWITCHING
US20180302940A1 (en) * 2014-03-28 2018-10-18 JVC Kenwood Corporation Transfer device, transfer method for transferring sound signals in different communication systems

Families Citing this family (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN102143544B (zh) * 2010-11-02 2014-03-05 华为技术有限公司 控制语音编码速率的方法、设备及系统
CN104703238A (zh) 2013-12-10 2015-06-10 华为技术有限公司 一种网络切换方法和终端设备
US10524179B2 (en) * 2015-04-10 2019-12-31 Qualcomm Incorporated Rate adaptation during handoffs
CN106576271B (zh) * 2015-05-30 2020-01-10 华为技术有限公司 语音编码速率调整的方法和设备
CN108924113A (zh) * 2018-06-25 2018-11-30 京信通信系统（中国）有限公司 语音编码速率的调整方法、装置、计算机存储介质及设备
CN109036444A (zh) * 2018-08-21 2018-12-18 京信通信系统（中国）有限公司 语音编码速率调整方法、装置、计算机存储介质及设备
CN113038534B (zh) * 2021-02-26 2023-04-07 海能达通信股份有限公司 窄带自组网中的呼叫打断方法及相关装置

Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1398126A (zh) * 2001-07-18 2003-02-19 华为技术有限公司 在通用移动电信系统中实现多种语音编解码的方法
CN101291538A (zh) * 2008-06-13 2008-10-22 华为技术有限公司 在基站收发信台进行通信交换的方法及基站子系统
EP2040420A1 (en) * 2007-09-20 2009-03-25 Lucent Technologies Inc. Method and device for shaping a stream of frames of user traffic, as well as computer network
CN101572910A (zh) * 2009-06-08 2009-11-04 华为技术有限公司 调整语音帧相位的方法、基站收发信台及基站子系统
CN102143544A (zh) * 2010-11-02 2011-08-03 华为技术有限公司 控制语音编码速率的方法、设备及系统

Family Cites Families (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20030210659A1 (en) * 2002-05-02 2003-11-13 Chu Chung Cheung C. TFO communication apparatus with codec mismatch resolution and/or optimization logic
WO2008086748A1 (en) * 2007-01-12 2008-07-24 Huawei Technologies Co., Ltd. A-interface-based mobile communication method,system and equipment
CN101222690B (zh) * 2008-02-01 2012-12-12 华为技术有限公司 一种编码切换方法、系统和设备

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1398126A (zh) * 2001-07-18 2003-02-19 华为技术有限公司 在通用移动电信系统中实现多种语音编解码的方法
EP2040420A1 (en) * 2007-09-20 2009-03-25 Lucent Technologies Inc. Method and device for shaping a stream of frames of user traffic, as well as computer network
CN101291538A (zh) * 2008-06-13 2008-10-22 华为技术有限公司 在基站收发信台进行通信交换的方法及基站子系统
CN101572910A (zh) * 2009-06-08 2009-11-04 华为技术有限公司 调整语音帧相位的方法、基站收发信台及基站子系统
CN102143544A (zh) * 2010-11-02 2011-08-03 华为技术有限公司 控制语音编码速率的方法、设备及系统

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2574100A4 (en) * 2010-12-28 2013-08-21 Huawei Tech Co Ltd FREQUENCY ADJUSTMENT METHOD AND DEVICE FOR TRFO LANGUAGE CALL SWITCHING
US20180302940A1 (en) * 2014-03-28 2018-10-18 JVC Kenwood Corporation Transfer device, transfer method for transferring sound signals in different communication systems
US10856351B2 (en) * 2014-03-28 2020-12-01 JVC Kenwood Corporation Transfer device, transfer method for transferring sound signals in different communication systems

Also Published As

Publication number Publication date
CN102143544B (zh) 2014-03-05
EP2635069A1 (en) 2013-09-04
CN102143544A (zh) 2011-08-03
EP2635069A4 (en) 2014-09-10

Similar Documents

Publication Publication Date Title
WO2012059051A1 (zh) 控制语音编码速率的方法、设备及系统
US11647428B2 (en) Communication terminal apparatus and communication method
RU2400937C2 (ru) СПОСОБ УПРАВЛЕНИЯ СКОРОСТЬЮ КОДЕКА, ЗАПРАШИВАЕМОГО УСТРОЙСТВОМ УПРАВЛЕНИЯ РЕСУРСОМ РАДИОСВЯЗИ ДЛЯ VoIP
KR100756092B1 (ko) 무선 통신 단말기 및 통신 방법
JP5956348B2 (ja) 可変レート・ボコーダを利用するユーザ機器のためのボイスオーバｉｐ容量を改善する方法
WO2009149636A1 (zh) 在基站收发信台进行通信交换的方法及基站子系统
CN107431569B (zh) 用于电路交换系统中的速率控制的方法和设备
WO2021180202A1 (zh) 一种辅助链路补偿通话的系统、装置及芯片
WO2008098490A1 (fr) Méthode et appareil de modification de codecs audio
EP2544506A1 (en) Method, device and system for voice communication
KR100559011B1 (ko) 가입자 기반 링백톤 서비스의 듀얼 코덱을 이용한 음질개선 방법
WO2017045125A1 (zh) 语音自适应参数的调整方法、系统及相关设备
WO2011009379A1 (zh) 一种下行非连续发射状态控制方法、装置及无线通信系统
CN106576271B (zh) 语音编码速率调整的方法和设备
US8396049B2 (en) Method and transcoder entity for tandem free operation in a telecommunication network
US10172046B2 (en) Mobile communication system, base station, higher-order apparatus, gateway apparatus, communication method, and program
WO2009046594A1 (fr) Procédé de négociation de codec entre un réseau sans fil et un réseau central dans un système de télécommunication mobile
WO2011069375A1 (zh) 控制本地交换的方法、装置和系统
EP2574100B1 (en) Rate adjustment method and apparatus applied to trfo voice call switching
CN103401833A (zh) 媒体编解码切换方法和装置
WO2011134354A1 (zh) 一种本地交换中实现切换的系统、方法及基站子系统
WO2010003451A1 (en) Method and transcoder entity for tandem free operation in a telecommunication network
WO2014205821A1 (zh) 语音通信方法、装置及系统
CN111757276B (zh) 一种语音格式的通知方法和装置
CN117041232A (zh) 一种语音通话实现方法、装置、设备、系统及存储介质

Legal Events

Date Code Title Description
121 Ep: the epo has been informed by wipo that ep was designated in this application

Ref document number: 11837577

Country of ref document: EP

Kind code of ref document: A1

NENP Non-entry into the national phase

Ref country code: DE

WWE Wipo information: entry into national phase

Ref document number: 2011837577

Country of ref document: EP