WO2009049536A1 - Procédé et dispositif pour la transmission de la voix dans un système radio - Google Patents

Procédé et dispositif pour la transmission de la voix dans un système radio Download PDF

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Publication number
WO2009049536A1
WO2009049536A1 PCT/CN2008/072628 CN2008072628W WO2009049536A1 WO 2009049536 A1 WO2009049536 A1 WO 2009049536A1 CN 2008072628 W CN2008072628 W CN 2008072628W WO 2009049536 A1 WO2009049536 A1 WO 2009049536A1
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WO
WIPO (PCT)
Prior art keywords
packet
voice
session
voice session
terminal
Prior art date
Application number
PCT/CN2008/072628
Other languages
English (en)
French (fr)
Inventor
Hang Li
Guanghan Xu
Yongquan Qiang
Hui Zhou
Qiang Ma
Original Assignee
Beijing Xinwei Telecom Technology Inc.
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 Beijing Xinwei Telecom Technology Inc. filed Critical Beijing Xinwei Telecom Technology Inc.
Priority to US12/682,518 priority Critical patent/US8331269B2/en
Publication of WO2009049536A1 publication Critical patent/WO2009049536A1/zh

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00Arrangements for detecting or preventing errors in the information received
    • H04L1/12Arrangements for detecting or preventing errors in the information received by using return channel
    • H04L1/16Arrangements for detecting or preventing errors in the information received by using return channel in which the return channel carries supervisory signals, e.g. repetition request signals
    • H04L1/18Automatic repetition systems, e.g. Van Duuren systems
    • H04L1/1829Arrangements specially adapted for the receiver end
    • H04L1/1835Buffer management
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L45/00Routing or path finding of packets in data switching networks
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L47/00Traffic control in data switching networks
    • H04L47/70Admission control; Resource allocation
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00Arrangements for detecting or preventing errors in the information received
    • H04L1/0001Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00Arrangements for detecting or preventing errors in the information received
    • H04L1/12Arrangements for detecting or preventing errors in the information received by using return channel
    • H04L1/16Arrangements for detecting or preventing errors in the information received by using return channel in which the return channel carries supervisory signals, e.g. repetition request signals
    • H04L1/1607Details of the supervisory signal
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L47/00Traffic control in data switching networks
    • H04L47/10Flow control; Congestion control
    • H04L47/26Flow control; Congestion control using explicit feedback to the source, e.g. choke packets
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L47/00Traffic control in data switching networks
    • H04L47/10Flow control; Congestion control
    • H04L47/34Flow control; Congestion control ensuring sequence integrity, e.g. using sequence numbers
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L47/00Traffic control in data switching networks
    • H04L47/70Admission control; Resource allocation
    • H04L47/76Admission control; Resource allocation using dynamic resource allocation, e.g. in-call renegotiation requested by the user or requested by the network in response to changing network conditions

Definitions

  • the present invention relates to the field of wireless communication technologies, and in particular, to a method and apparatus for transmitting voice in a wireless communication system. Background technique
  • resources between a terminal and a base station may be shared by multiple services of the terminal, and the base station adopts according to service QoS and link quality.
  • the scheduling method dynamically allocates resources for various services. Since the existing VOIP mostly uses RTP/RTCP voice encapsulation based on the UDP transport protocol to provide end-to-end real-time voice transmission, this transmission mode will lead to low-cost speech coding such as the G.729 series. Transmission efficiency, which is a great waste of air resources for wireless systems. Thus, special packaging for voice packets and proper allocation of air resources become important.
  • the voice service resource may be occupied by a higher priority air interface link control message (such as a resource configuration change message), so that the voice packet of the current frame cannot be sent in time.
  • a higher priority air interface link control message such as a resource configuration change message
  • Embodiments of the present invention provide a method and apparatus for transmitting voice in a wireless system to ensure real-time transmission of voice services and increase reliability of voice transmission.
  • Embodiments of the present invention provide a method for transmitting voice in a wireless system, including the following steps:
  • the sender sends the original voice coded packet identifier to be used to identify the sequence number of the play sequence, and performs channel coding on the voice coded packet after the identifier sequence to form a voice session packet;
  • the sending end establishes a voice session or a voice data mixing session with the receiving end, and dynamically allocates a channel for the voice session or the voice data mixing session;
  • the sending end sends the newly arrived voice session packet, the delayed voice session packet, the retransmitted voice session packet, the data session packet, and the control command packet according to the preset priority;
  • the receiving end receives the voice session packet and detects the voice session packet. If it confirms that the voice session packet is lost, it sends a missing voice session packet number to the sender.
  • the NACK packet is used to instruct the transmitting end to retransmit the voice session packet. If the receiving end is the terminal, the method further includes: placing the correctly received voice session packet into the jitter buffer controller of the receiving end.
  • Embodiments of the present invention also provide an apparatus for transmitting voice in a wireless system, the apparatus comprising:
  • the packet encapsulation unit is installed at the base station and the terminal, and the original voice coding packet to be sent is used to identify the sequence number of the play sequence, and the voice coding packet after the identification sequence number is channel coded to form a voice conversation packet;
  • Resending a voice session packet generator installed at the base station and the terminal, generating a voice conversation packet that needs to be retransmitted;
  • a channel allocation unit installed at the base station, interacting with the terminal to establish a voice session establishment request message to establish a voice session or a voice data hybrid session, and dynamically allocate a channel;
  • the sending packet scheduling unit is installed at the base station and the terminal, and is configured to, according to a preset priority, a newly arrived voice session packet, a delayed voice session packet, a retransmission voice session packet, and a data conference.
  • the voice packet and the control command packet are sent;
  • the voice packet detecting unit is installed at the base station and the terminal, receives the voice session packet and detects the voice session packet, and if it confirms that the voice session packet is lost, notifies the NACK packet generator that the lost voice session packet is confirmed; The voice packet detecting unit sends the correctly received voice conversation packet to the jitter buffer controller;
  • the NACK packet generator is installed at the base station and the terminal, generates a NACK packet containing the lost voice conference packet sequence number, and sends a NACK packet to the sender transmitting the lost voice session packet to instruct the sender to retransmit the voice session packet.
  • the real-time service QoS requirement can be met.
  • FIG. 1 is a schematic diagram of a voice transmission model in a conventional wireless communication system
  • FIG. 2 is a schematic diagram of a voice transmission model in a typical all-IP service wireless communication system
  • FIG. 3 is a schematic diagram of a voice transmission model in an improved all-IP service wireless communication system according to an embodiment of the present invention
  • FIG. 4 is a schematic diagram of an improved voice session packet transmission format and transmission process according to an embodiment of the present invention.
  • FIG. 5 is a schematic diagram of an improved voice session packet transmission and dynamic channel allocation according to an embodiment of the present invention.
  • 6 is a schematic diagram of an improved voice tongue packet transmission apparatus according to an embodiment of the present invention
  • 7 is a flowchart of an overall process of an improved voice session packet transmission method according to an embodiment of the present invention
  • FIG. 8 is a flow chart of signaling establishment of an embodiment of the present invention.
  • FIG. 9 is a flowchart of a voice packet sent by a transmitting end according to an embodiment of the present invention.
  • FIG. 10 is a flowchart of receiving a voice session packet by a receiving end according to an embodiment of the present invention.
  • FIG. 11 is a schematic diagram of a jitter buffer controller according to an embodiment of the present invention. ;
  • FIG. 12 is a flowchart of retransmission indication signaling according to an embodiment of the present invention. Mode for carrying out the invention
  • each information bitstream coding block is converted into a signal waveform and placed in a frame for transmission.
  • a fixed channel is usually used to transmit the voice coding package.
  • the channel may adopt a fixed code modulation mode and adapt to the channel change by controlling the transmit power.
  • one voice packet per frame periodically arrives at the transmitting end. Each frame is strictly transmitted through the air channel, and the transmission order and playback order of the packets must be the same.
  • 1 is a schematic diagram of a voice transmission model in a conventional wireless communication system, showing an example of downlink voice transmission.
  • the packet from the vocoder is passed to the base station via the radio network controller and utilizing the T1 link.
  • the time and order of arrival of the package is fixed.
  • the wireless channel may cause a packet error.
  • a packet with a number of 2 is detected as a bit error when the packet is transmitted over the air interface, and the packet is discarded. Such as If voice packets are dropped frequently, the voice quality will drop.
  • each voice packet is packaged into an IP voice packet. Since the existing VOIP systems mostly use RTP/RTCP voice encapsulation based on the UDP transport protocol to provide end-to-end real-time voice transmission, the overhead caused by this transmission method is low-rate speech coding such as the G.729 series. Said that will lead to degraded transmission efficiency.
  • Figure 2 is a schematic diagram of a voice transmission model in a typical all-IP service wireless communication system. In Figure 2, each IP voice packet may introduce additional delay jitter when transmitted over the network. In the radio access network, IP voice packets are transmitted as data services over the air interface.
  • FIG. 3 is a schematic diagram of a voice transmission model in an improved all-IP service wireless communication system according to an embodiment of the present invention.
  • the wireless communication system is still an all-IP wireless system.
  • the IP voice packet Once the IP voice packet enters the wireless access system, the IP header of the IP voice packet is removed, and only the voice session packet that is specially encapsulated by the original voice code packet is transmitted through the air interface. At this point, voice packets can be transmitted with smaller wireless resources.
  • Figure 3 shows an improved IP voice packet transmission system. In order to ensure the voice quality under the condition of channel fading, each original speech coding packet is added with an ID serial number, and the retransmission of the speech encoded packet is realized by using such an ID serial number.
  • the voice coding packet with each ID number is channel-coded to form a voice conversation packet, and the ratio of the original voice coding packet length to the voice conversation packet length is not less than 0.70.
  • Receiving The terminal will detect if the voice conversation packet is received correctly. If the voice session packet is lost due to channel fading, the ID sequence number of the voice session packet will be fed back to the sender to request retransmission. It is possible to detect multiple voice session packets lost, at this time, all lost voice session packets
  • the ID sequence number will be fed back to the sender to request retransmission of multiple voice session packets.
  • FIG. 4 is a schematic diagram of an improved voice session packet transmission format and transmission process according to an embodiment of the present invention.
  • each G.729 speech coding packet is appended with an ID sequence number and a CRC calibration before transmission, and channel coding is performed for each speech coding packet with the ID sequence number.
  • channel coding is performed for each speech coding packet with the ID sequence number.
  • the NACK message containing the ID number of the voice session packet is sent back to the sender.
  • the sender receives the NACK message before sending the 4th packet, and the sender will automatically send the 2nd and 4th packets at the same time, so that no permanent delay will be introduced.
  • the receiving end further includes a jitter buffer controller, which mainly controls the playing delay of each voice session packet and performs necessary reordering of the received voice session packets, so that the output voice encoding packets can be in the correct order and normal.
  • the speed of speech has been played.
  • one or more synthesized speech coding packets need to be inserted when the output speech coding packet is missing, and one or more speech coding packets need to be discarded when the output speech coding packet is congested.
  • All-IP wireless systems support mixed data and voice transmissions, so each session may require the transmission of voice session packets and data session packets.
  • the available bandwidth is always prioritized for sending voice packets.
  • the ID number of the first voice session packet needs to be sent at the beginning of the voice session or voice data mixing session. Give the other end, so that the other end can correctly detect the received voice session packet, and initiate retransmission once the voice session packet is confirmed to be lost.
  • the voice signal packet loss is confirmed by using the received signal to noise ratio or the error detection code check.
  • the optimization method is to adjust the number of allocated channels in real time, and the channel position and its corresponding modulation and coding mode make the communication channel in optimal condition.
  • the base station needs to send a bandwidth reconfiguration command to the terminal.
  • the base station sends a bandwidth reconfiguration command packet to the terminal to increase the uplink or downlink bandwidth, and the increased bandwidth is automatically cancelled after the retransmitted voice session packet is sent, and the base station does not need to additionally send the control command packet to the terminal.
  • the base station transmits a reconfiguration command to the terminal by using a channel allocated to the service packet, which is called in-band signaling.
  • the service packet includes only voice session packets
  • one or more voice session packets must be delayed in time to transmit signaling messages. Since the reconfiguration command needs to be sent periodically, there is a delay in the arrival time of the voice session packet at the receiving end.
  • the transmission of the voice packet is interrupted for transmitting a NACK message
  • the interrupted voice session packet will be delayed and transmitted with the newly arrived voice session packet.
  • FIG. 5 is a schematic diagram of voice tongue packet transmission and dynamic channel allocation in a TDD wireless communication system according to an embodiment of the present invention.
  • the base station and the terminal respectively use the downlink frame and the uplink frame to transmit the voice session packet, and each uplink or downlink voice session packet is identified with a sequence number. The base station needs to send according to the current frame.
  • the voice session packet and the data session packet and the current channel condition to dynamically allocate the number of channels, the channel location, and the modulation and coding mode corresponding to each channel occupied by the voice session packet and the data session packet, wherein the voice session packet to be sent may include a new one.
  • the base station sends a bandwidth reconfiguration command to the terminal after detecting the loss of the uplink voice session packet, the command reaching the terminal no later than the NACK packet containing the lost voice session packet number. For example, if the second downlink voice session packet D2 is lost, a NACK packet is sent to the base station in the subsequent uplink frame.
  • the base station sends a bandwidth reconfiguration (BWR) command to the terminal to notify the terminal to reconfigure the channel, so that the terminal can send two voice conversation packets U2 and U3 in the next uplink frame, and the next downlink
  • BWR bandwidth reconfiguration
  • the frame is capable of transmitting three voice conversation packets D2, D3 and D4.
  • the reconfiguration is only valid for one frame, and the channel will automatically return to the previous bandwidth state after reconfiguration.
  • the base station will immediately send a NACK packet to the terminal, and trigger the terminal to retransmit the voice conversation packet U4.
  • the base station will also send a BWR command to the terminal to initiate reconfiguration, so that voice session packets D5, D6 and D7 will be sent in one downlink frame; voice packets U5 and U6 will be sent in one upstream frame. .
  • reconfiguration is only valid for one frame.
  • the terminal's jitter buffer controller can adjust the playback delay so that the voice coded packet is played at the normal speed consistent with the vocoder.
  • FIG. 6 is a schematic diagram of an improved voice tongue packet transmission apparatus according to an embodiment of the present invention.
  • the packet encapsulating unit 600 identifies the original speech coding packet identification number to characterize its playing order, and performs channel coding on the speech encoded packet of the identification serial number to be converted into a voice conversation packet.
  • the retransmission voice session packet generator 603 generates a voice conversation packet to be retransmitted.
  • Letter The channel allocating unit 601 allocates the number of channels for packet transmission according to the condition of the packet waiting in the transmission buffer, and interacts with the terminal to establish voice session establishment request information to establish a voice session or a voice data mixing session, and also generates a reconfiguration to be transmitted to the terminal. Command package.
  • the channel allocating unit determines the sequence number of the first voice session packet according to the starting sequence number of the voice packet carried in the voice session establishment request message that the terminal interacts with the base station. Moreover, the channel allocation unit in the base station dynamically allocates the number of channels, the channel location, and the modulation and coding mode corresponding to each channel occupied by the voice session packet and the data session packet according to the voice session and the data session packet to be sent in the current frame and the current channel condition.
  • the voice session packet to be sent includes a newly arrived voice session packet, a delayed voice session packet, or a voice conversation packet to be retransmitted.
  • the channel allocation unit of the base station also sends a bandwidth reconfiguration command to the terminal to increase the uplink or downlink bandwidth.
  • the increased bandwidth is automatically cancelled after the retransmitted voice session packet is sent, and the base station does not need to additionally send a control command packet to the terminal.
  • the NACK packet generator 602 generates a NACK packet containing the lost voice session packet number after the voice packet detecting unit 606 detects the loss of the voice session packet from the terminal.
  • the newly arrived voice session packet and the delayed voice session packet generated by the packet encapsulation unit 600, the reconfiguration command packet generated by the channel allocation unit 601, the NACK packet generated by the NACK generator, and the retransmission voice session generated by the retransmission voice session packet generator The packets are all sent to the sending packet scheduling unit 604, and the sending packet scheduling unit 604 selects one or more packets to be sent according to the predetermined priority and available bandwidth conditions, and the priority is from high to low in the order of control command packets, such as With command packets and NACK packets, retransmission voice session packets, delayed voice session packets, new arrival voice session packets, data session packets.
  • the voice packet detecting unit 606 receives the voice session packet from the terminal and detects, and notifies the NACK packet generator 602 of the voice session packet that has been confirmed to be lost.
  • a jitter buffer controller 605 may also be further included as needed for detecting a voice packet from the voice packet.
  • Element 606 receives the correct voice conversation packet.
  • the packet encapsulating unit 607, the retransmission voice session packet generator 608, the NACK packet generator 609, the voice packet detecting unit 611, and the transmission packet scheduling unit 610 process the voice session packet from the base station, and each unit function and the base station The side is similar.
  • the above method proposes a reliable voice session packet transmission mechanism at the expense of delay and delay jitter, so at least one jitter buffer controller 612 is to be configured on the terminal side.
  • the jitter buffer controller 612 receives the correct voice session packets from the voice packet detection unit 611 and adjusts the playback delay for each voice session packet. Then, the output voice conference packet is sent to the speech encoding packet extracting unit for decapsulation, and the original speech encoding packet is generated for the speech encoder.
  • FIG. 7 is a flowchart of an overall process of an improved voice session packet transmission method according to an embodiment of the present invention. As shown in FIG. 7, the method includes the following steps:
  • Step 100 The application layer of the wireless communication system receives the original voice coded packet and adds the sequence number in order, and performs channel coding on the original voice coded packet with the added sequence number to encapsulate the voice session packet.
  • Step 101 In the voice session establishment process, the session establishment request message carries the starting sequence number of the voice session packet, so that the two parties unify the sequence number of the first voice session packet, and the voice session establishment process is as shown in FIG. 8.
  • the calling party for example, the sending end in this embodiment
  • the called party for example, the receiving end in this embodiment
  • the voice session establishing request message carries the voice session.
  • the starting sequence number of the packet, the called party returns the voice session establishment response to the calling party, thereby establishing a voice tongue.
  • Step 102 After the voice session is successfully established, the sending end sends the step to the receiving end frame by frame.
  • the process of transmitting the voice conversation packet by the voice end tongue packet generated in 100 is as shown in FIG. 9.
  • FIG. 9 is a flowchart of sending a voice conversation packet by a sending end according to an embodiment of the present invention, which includes the following steps 301-304.
  • Step 301 The sender adds the voice session packet to the sending packet scheduling unit.
  • Step 302 The sending packet scheduling unit determines whether there is a data session packet with a higher priority than the voice session packet. If there is no data session packet with a higher priority than the voice session packet, step 303 is performed; if there is a higher priority than the voice session packet In the case of the data session packet, step 304 is performed.
  • Step 303 The sending packet scheduling unit sends the current voice session packet to the receiving end, and thus, the sending end completes the normal sending process of the voice session packet.
  • Step 304 The sending packet scheduling unit buffers the voice session packet, first sends a data session packet with a higher priority than the voice conference packet, and then sends the current voice session packet to the receiving end according to the bandwidth status.
  • Step 103 The receiving end receives the voice session packet sent by the sending end, and identifies the voice session packet that is not received and received incorrectly, and generates a NACK packet containing the voice session packet number that is not received and received incorrectly and feeds back to the sending. end.
  • the process of receiving the voice conversation packet at the receiving end is shown in Figure 10.
  • FIG. 10 is a flowchart of receiving a voice packet by a receiving end according to an embodiment of the present invention, which includes the following steps 401-404.
  • Step 401 The voice packet detecting unit at the receiving end determines whether the current frame correctly receives the voice session packet sent by the sending end. If the voice session packet is correctly received, step 402 is performed; otherwise, step 403 is performed.
  • Step 402 The voice packet detecting unit at the receiving end puts the correctly received voice session packet into the jitter buffer controller shown in FIG. 11.
  • the specific placing rule is: performing reordering function according to the sequence number of the arrived voice session packet, and calculating The buffer position Buf_In_p corresponding to the serial number (range 0 ⁇ Max_Buf_Size-1) is placed in the buffer position indicated by the Buf_In_p and executed. Step 404.
  • Step 403 If the voice session detecting unit does not correctly receive the voice session packet, the voice session packet in the buffer location indicated by the Buf_In_p corresponding to the sequence number of the voice session packet is lost.
  • Step 104 The jitter buffer controller at the receiving end calculates an optimal delay N d between the playing time and the arrival time of each voice session packet, such as when the relationship between Buf_In_p and Buf_Out_p is detected.
  • N d the delay reaches the N d frame
  • the jitter buffer controller starts the sending operation, and submits the voice session packet to the voice coding packet extracting unit according to the sequence number indicated by the Buf_Out_p, and each time a voice conversation packet is submitted, Buf_Out_p is automatically Add a unit whose range is 0 ⁇ Max—Buf—Size-1.
  • Step 105 The voice encoding packet extracting unit at the receiving end receives the voice session packet, and decapsulates and generates the original voice encoding packet.
  • the speech encoding packet extracting unit uniformly outputs the original speech encoding packet to the application layer playback according to the specified speech encoding rate.
  • the speech encoding packet extracting unit may also insert one or more synthesized speech encoding packets according to the sequence number when the output speech encoding packet is missing or delete one or more speech encoding packets when the output speech encoding packet is congested.
  • Step 106 After receiving the NACK message, the voice packet detecting unit of the sending end determines whether the current bandwidth resource is sufficient to transmit all the retransmitted voice session packets and the current frame voice session packet, and if yes, sends all the retransmitted voice session packets and the current frame.
  • the voice session packet otherwise, the media access layer is requested to perform the resource reconfiguration process, and after the configuration is successful, the retransmission voice session packet and the newly arrived voice session packet are sent.

Description

无线系统中传输语音的方法及装置
技术领域
本发明涉及无线通信技术领域, 特别涉及一种无线通信系统中传输 语音的方法及装置。 背景技术
在基于 IP的多业务混合传输的无线通信系统中,为了合理利用无线 资源, 一个终端与基站间的资源可能会由该终端的多种业务共享, 基站 根据业务的 QoS 及链路质量等情况采用调度的方式动态地为各种业务 分配资源。 由于现有的 VOIP大多采用基于 UDP传输协议的 RTP/RTCP 语音封装, 以提供端到端的实时语音传输, 所以这种传输方式对于低码 率语音编码如 G.729系列来说将导致艮低的传输效率, 而对于无线系统 而言这种低频语效率是对空中资源的极大浪费。 因而, 对于语音包的特 殊封装及空中资源的合理分配变得十分重要。 在这种情况下, 语音业务 的资源有可能会被优先级更高的空口链路控制消息(如资源配置改变消 息) 占用, 导致当前帧的语音包无法及时 送。 这样对于实时性要求 较高的语音业务将产生影响。 此外, 由于无线信道的不可靠性, 会导致 接收端语音包的错误接收或丢失。 发明内容
本发明的实施例提出了一种无线系统中传输语音的方法及装 置, 以保证语音业务的实时传输, 并增加语音传输的可靠性。
本发明的实施例提出了一种无线系统中传输语音的方法, 包括 以下步骤: 发送端对需发送的原始语音编码包标识用于表征播放顺序的 序号, 并对标识序号后的语音编码包进行信道编码构成语音会话 包;
发送端与接收端建立语音会话或语音数据混合会话, 为所述语 音会话或语音数据混合会话动态分配信道;
发送端根据预设的优先级对新到达语音会话包、 延迟语音会话 包、 重传语音会话包, 数据会话包及控制命令包进行发送;
接收端接收语音会话包并对所述语音会话包进行检测, 如果确 认语音会话包丢失, 向发送端发送包含丢失的语音会话包序号的
NACK 包以指示发送端重传该语音会话包; 如果所述接收端为终 端, 进一步包括: 将正确接收到的语音会话包放入接收端的抖动緩 冲控制器。
本发明的实施例还提出一种无线系统中传输语音的装置, 该装 置包括:
包封装单元, 安装于基站和终端, 对需发送的原始语音编码包 标识用于表征播放顺序的序号, 并对标识序号后的语音编码包进行 信道编码构成语音会话包;
重发语音会话包产生器, 安装于基站和终端, 生成需要重传的 语音会话包;
信道分配单元, 安装于基站, 与终端交互语音会话建立请求消 息以建立语音会话或语音数据混合会话 , 动态分配信道;
发送包调度单元, 安装于基站和终端, 用于根据预设的优先级 对新到达语音会话包、 延迟语音会话包、 重传语音会话包、 数据会 话包及控制命令包进行发送;
语音包检测单元, 安装于基站和终端, 接收语音会话包并对所 述语音会话包进行检测, 如果确认语音会话包丢失, 将确认丢失的 语音会话包通知给 NACK包产生器;安装于终端的语音包检测单元 将正确收到的语音会话包送入抖动緩冲控制器;
NACK包产生器, 安装于基站和终端, 生成包含丢失的语音会 话包序号的 NACK包,并向发送该丢失的语音会话包的发送端发送 NACK包以指示发送端重传该语音会话包。
由此可见, 在本发明实施例提出的传输语音的方法和装置中, 通过 针对特殊封装的语音包引入重传机制、 动态信道分配和延迟抖动緩冲机 制,可在满足实时性业务 QoS要求的 上,提高多业务传输的无线系 统中实时性语音业务的频语效率和可靠性。 附图简要说明
图 1为传统的无线通信系统中语音传输模型的示意图; 图 2为典型的全 IP业务无线通信系统中语音传输模型的示意 图;
图 3为本发明实施例的改进的全 IP业务无线通信系统中语音传 输模型的示意图;
图 4为本发明实施例的改进的语音会话包传输格式和传输过程 的示意图;
图 5为本发明实施例的改进的语音会话包传输和动态信道分配 的示意图;
图 6为本发明实施例的改进的语音^舌包传输装置示意图; 图 7为本发明实施例的改进的语音会话包传输方法的总体处理 流程图;
图 8为本发明实施例的^舌建立信令流程图;
图 9为本发明实施例的发送端发送语音^舌包流程图; 图 10为本发明实施例的接收端接收语音会话包流程图; 图 11为本发明实施例的抖动緩冲控制器的示意图;
图 12为本发明实施例的重传指示信令流程图。 实施本发明的方式
为了使本发明的目的、 技术方案及优点更加清楚明白, 以下结合附 图及实施例,对本发明的具体实施方式进行进一步详细说明。应当理解, 此处所描述的具体实施例仅仅用以解释本发明 , 并不用于限定本发明。
在无线通信系统中, 每个信息比特流编码块都被转换为信号波 形并放入一个帧中进行传输。 由于语音业务固有的实时性和带宽恒 定性, 因此在传统无线通信系统的语音传输方式中, 通常使用固定 的信道来传输语音编码包。 该信道可能采用固定的编码调制方式, 并通过控制发射功率的方式来适应信道的变化。 在传统的无线通信 系统中, 每帧有一个语音包周期性地到达发送端。 每帧通过空中信 道严格传送一个语音包, 包的传输顺序和播放顺序必须相同。 图 1 为传统的无线通信系统中语音传输模型的示意图, 示出了一个下行 语音传输的例子。 由图 1可见, 通过无线网络控制器并利用 T1链 路将来自声码器的包传递到基站。 包到达的时间和顺序是固定不变 的。 然而, 无线信道可能引起包错误。 在图 1中, 包通过空中接口 传输时检测出号码为 2的包发生了比特错误, 于是该包被丢弃。 如 果频繁的丢弃语音包, 语音质量将会下降。
对于一个典型的全 IP无线系统,在核心网中,每个语音包都被 打包成一个 IP语音包。 由于现有的 VOIP系统大多采用基于 UDP 传输协议的 RTP/RTCP语音封装, 以提供端到端的实时语音传输, 所以这种传输方式带来的额外开销对于低码率语音编码如 G.729系 列来说将导致艮低的传输效率。图 2为典型的全 IP业务无线通信系 统中语音传输模型的示意图。在图 2中,每个 IP语音包在网络中传 输时都可能会引入额外的时延抖动。 在无线接入网中, IP语音包作 为数据业务通过空中接口传输。 对于空中丢包, 基于无线数据传输 系统采用的自动重传请求机制 (ARQ )和 IP语音包播放时采用的 抖动緩冲控制,可以获得较高的语音质量,但是由于 IP语音包头引 起的开销, 较高的语音质量是以较高的无线资源损耗为代价的。 图 2所示的就是这样的一个 IP语音包传输系统。
图 3为本发明实施例的改进的全 IP业务无线通信系统中语音传 输模型的示意图。 在本发明的这一实施例中, 无线通信系统仍然为 一个全 IP无线系统。 一旦 IP语音包进入无线接入系统, 将 IP语音 包的 IP头去掉,通过空中接口只传输对原始语音编码包经过再次特 殊封装而成的语音会话包。 此时, 可以用较小的无线资源传输语音 包。 图 3所示的就是一个这样的改进的 IP语音包传输系统。为了保 证在信道衰落条件下的语音质量, 每个原始语音编码包都加上一个 ID序号, 利用这样的 ID序号实现语音编码包的重传。 之后, 再对 每个加上 ID序号后的语音编码包进行信道编码从而构成语音会话 包, 原始语音编码包长度与语音会话包的长度比不小于 0.70。 接收 端将检测是否正确接收语音会话包。 如果语音会话包由于信道衰落 而丢失, 该语音会话包的 ID序号将被反馈给发送端以请求重传。 可能检测到多个语音会话包丢失, 此时, 所有丢失的语音会话包的
ID序号将被反馈给发送端以请求重传多个语音会话包。
图 4为本发明实施例的改进的语音会话包传输格式和传输过程 示意图。 在本发明的这一实施例中, 给每个 G.729语音编码包在传 输前都加上一个 ID序号和 CRC校 ^r, 并对每个加上 ID序号后的 语音编码包进行信道编码从而构成语音会话包。 在接收端, 检测到 第 2个语音会话包丢失了,则含有该语音会话包的 ID序号的 NACK 消息被发送回发送端。发送端在发送第 4个包前接收到 NACK消息, 则发送端将自动同时发送第 2个和第 4个包, 这样就不会引入永久 时延。 接收端还包括一个抖动緩冲控制器, 主要控制每个语音会话 包的播放时延并对接收到的语音会话包进行必要的重新排序, 这样 输出的语音编码包就能以正确的顺序和正常的语速播放了。 为了保 持正常的播放语速, 在输出语音编码包缺失时需要插入一个或多个 合成语音编码包, 在输出语音编码包拥塞时需要丢弃一个或多个语 音编码包。
全 IP无线系统支持数据和语音混合传输,因此每一帧都可能需 要传输语音会话包和数据会话包。 在本发明的一个实施例中, 如果 没有足够的带宽来发送所有数据, 可用带宽总是优先用来发送语音 ^舌包。
为了使发送端和接收端之间的语音传输同步, 在语音会话或语 音数据混合会话开始时, 需要将第一个语音会话包的 ID序号发送 给另一端, 使得另一端能够正确检测接收到的语音会话包, 一旦确 认语音会话包丢失就初始化重传。 在本发明实施例中, 利用接收信 噪比或检错码校验来确认语音会话包丢失。
由于在实际的信道中, 每一信道所受到的干扰和传播特性会发 生变化, 优化的作法是实时调整分配的信道数, 信道位置及其相应 的调制和编码方式使得通信信道处于最佳条件。 为了达到这个目 的, 基站需要向终端发送带宽重配命令。 基站向终端发送带宽重配 命令包以增加上行或下行带宽, 该增加的带宽在重传的语音会话包 发送后自动取消, 无需基站向终端额外发送控制命令包。 在本发明 的一个实施例中, 基站利用分配给业务包的信道将重配命令发送给 终端, 称为带内信令。 如果业务包仅包括语音会话包, 则必须延迟 一个或多个语音会话包以及时传输信令消息。 由于重配命令需要周 期地发送, 因此, 在接收端, 语音会话包的到达时间会出现延迟。 当语音^舌包的传输被中断用来传输 NACK消息时,被中断的语音 会话包将被延迟, 并且和新到达的语音会话包一起传输。 在以上的 情形中, 为了避免插入控制命令而引入永久时延, 在重配并发送该 帧中的延迟语音会话包和新到达语音会话包时, 需要将所需的额外 带宽计算在内以进行补偿。 如果业务包既包括语音会话包又包括数 据^舌包, 当需要传输重配命令或 NACK消息时,将中断数据 ^舌 包以释放带宽用于传输命令。 图 5为本发明实施例的 TDD无线通 信系统中的语音^舌包传输和动态信道分配的示意图。 在图 5中, 基站和终端分别利用下行帧和上行帧来传输语音会话包, 并且给每 个上行或下行语音会话包都标识一个序号。 基站根据当前帧需发送 的语音会话包及数据会话包和当前信道条件来动态分配信道数目、 信道位置及语音会话包和数据会话包所占用的每一信道对应的调 制编码方式, 其中需发送的语音会话包可以包括新到达的语音会话 包、 被延迟的语音会话包以及要重传的语音会话包的其中之一或任 意组合。 而且, 基站在检测到上行语音会话包丢失后向终端发送带 宽重配命令,该命令不迟于包含丢失的语音会话包序号的 NACK包 到达终端。例如,如果第 2个下行语音会话包 D2丢失,一个 NACK 包在随后的上行帧中被发送给基站。一旦基站接收到 NACK包,便 发送一个带宽重配( BWR )命令给终端用于通知终端重配信道, 这 样终端就能在下一个上行帧时发送 U2和 U3两个语音会话包了, 下一个下行帧能够发送三个语音会话包 D2、 D3和 D4。 另外, 重 配仅对一帧有效, 在重配后信道将自动返回前一个带宽状态。 当第 四个上行语音会话包 U4丢失, 基站将立即发送 NACK包给终端, 并触发终端重传语音会话包 U4。 在下一个下行帧中, 基站也将发 送 BWR命令给终端以初始化重配, 这样语音会话包 D5, D6和 D7 将在一个下行帧中发送; 语音^舌包 U5和 U6将在一个上行帧中 发送。 另外, 重配仅对一个帧有效。 显然, 上述传输方案引入了语 音会话包到达时间的抖动。 终端的抖动緩冲控制器可以对播放时延 进行调整, 从而使语音编码包以与声码器一致的正常速度播放。
图 6为本发明实施例的改进的语音^舌包传输装置示意图。 在 基站侧, 包封装单元 600将原始语音编码包标识序号以表征其播放 顺序, 并对标识序号的语音编码包进行信道编码从而转换为语音会 话包。 重发语音会话包产生器 603生成将要重传的语音会话包。 信 道分配单元 601根据传输緩存中等待的包的情况分配用于包传输的 信道个数, 并与终端交互语音会话建立请求信息以建立语音会话或 语音数据混合会话, 还生成发送给终端的重配命令包。 信道分配单 元根据终端与基站交互的语音会话建立请求消息中携带的语音包 的起始序号, 来确定第一个语音会话包的序号。 而且, 基站中的信 道分配单元根据当前帧需发送的语音会话及数据会话包和当前信 道条件动态分配信道数目、 信道位置及语音会话包和数据会话包所 占用的每一信道对应的调制编码方式, 其中需发送的语音会话包包 括新到达的语音会话包、 被延迟的语音会话包或要重传的语音会话 包。 基站的信道分配单元还向终端发送带宽重配命令以增加上行或 下行带宽, 增加的带宽在重传的语音会话包发送后自动取消, 无需 基站向终端额外发送控制命令包。 NACK包产生器 602在语音包检 测单元 606检测到来自终端的语音会话包丢失后生成包含丢失的语 音会话包序号的 NACK包。包封装单元 600生成的新到达的语音会 话包和延迟语音会话包、 信道分配单元 601 生成的重配命令包、 NACK产生器生成的 NACK包和重发语音会话包产生器生成的重 传语音会话包都被发送到发送包调度单元 604,发送包调度单元 604 根据预定的优先级和可用带宽情况来选择一个或多个包进行发送, 优先级由高到低的顺序为控制命令包, 如重配命令包和 NACK包、 重传语音会话包、 延迟语音会话包、 新到达语音会话包、 数据会话 包。 语音包检测单元 606接收来自终端的语音会话包并检测, 将确 认丢失的语音会话包通知给 NACK包产生器 602。 在基站侧, 也可 以根据需要进一步包括抖动緩冲控制器 605 , 用于从语音包检测单 元 606接收正确的语音会话包。 在终端侧, 包封装单元 607、 重发 语音会话包产生器 608、 NACK包产生器 609、语音包检测单元 611 和发送包调度单元 610对来自基站对语音会话包进行处理, 各单元 功能与基站侧类似。
上述方法提出了一种以延迟和延迟抖动为代价的可靠的语音 会话包传输机制, 因此在终端侧至少要配置一个抖动緩冲控制器 612。 抖动緩冲控制器 612从语音包检测单元 611接收正确的语音 会话包, 并为每个语音会话包调整播放延迟。 然后, 输出的语音会 话包被发送到语音编码包提取单元进行去封装, 为语音编码器生成 原始语音编码包。
图 7为本发明实施例的改进的语音会话包传输方法的总体处理 流程图, 如图 7所示, 该方法包括如下步骤:
步骤 100, 无线通信系统的应用层接收原始语音编码包并按顺 序加入序号, 并对加入序号的原始语音编码包进行信道编码而封装 成语音会话包。
步骤 101 , 在语音会话建立过程中, 在会话建立请求消息中携 带语音会话包的起始序号, 以使双方统一第一个语音会话包的序 号, 语音会话建立过程如图 8所示。 如图 8所示, 主叫方(例如本 实施例中的发送端)向被叫方(例如本实施例中的接收端)发起语 音会话建立请求消息, 该语音会话建立请求消息中携带语音会话包 的起始序号, 被叫方向主叫方返回语音会话建立响应, 从而建立语 音^舌。
步骤 102, 语音会话建立成功后, 发送端逐帧向接收端发送步骤 100中生成的语音^舌包, 发送端发送语音会话包的过程如图 9所 示。
图 9 为本发明实施例的发送端发送语音会话包的流程图, 包括 如下步骤 301 - 304。
步骤 301 , 发送端将语音会话包存入发送包调度单元中。
步骤 302, 发送包调度单元判断是否有比语音会话包优先级高 的数据会话包发送, 若没有比语音会话包优先级高的数据会话包, 执行步骤 303; 若有比语音会话包优先级高的数据会话包时, 则执 行步骤 304。
步骤 303, 发送包调度单元发送当前语音会话包到接收端, 至 此, 发送端完成语音会话包的正常发送流程。
步骤 304, 发送包调度单元緩存语音会话包, 先发送比语音会 话包优先级高的数据会话包, 再根据带宽状况发送当前语音会话包 到接收端。
步骤 103, 接收端接收发送端发送的语音会话包, 对没有收到 的及错误接收的语音会话包进行标识, 生成包含没有收到的及错误 接收的语音会话包序号的 NACK包并反馈给发送端。接收端接收语 音会话包的过程如图 10所示。
图 10为本发明实施例的接收端接收语音^舌包的流程图, 包括 如下步骤 401 - 404。
步骤 401 , 接收端的语音包检测单元判断当前帧是否正确接收 到发送端发送的语音会话包, 如果正确接收到语音会话包, 执行步 骤 402; 否则, 执行步骤 403。 步骤 402, 接收端的语音包检测单元将正确接收的语音会话包 放入图 11 所示的抖动緩冲控制器中, 具体放置规则为: 根据到达 的语音会话包的序号, 执行重排序功能, 计算出该序号对应的緩冲 器位置 Buf— In— p (范围为 0~Max— Buf— Size-1 ), 将语音会话包放入 该 Buf— In— p指示的緩冲器位置中, 并执行步骤 404。
步骤 403,如果语音会话检测单元没有正确接收到语音会话包, 标识该语音会话包的序号对应的 Buf— In— p 指示的緩冲器位置中的 语音会话包为丢失。
步骤 404 , Buf— In— p 自动增加一个单位, 当 Buf— In— p= Max— Buf— Size时, 使 Buf— In— p=0。
步骤 104, 接收端的抖动緩冲控制器计算每一语音会话包的播 放时间与到达时间之间的最佳时延 Nd , 如检测到 Buf— In— p 与 Buf— Out— p之间的时延达到 Nd帧时,抖动緩冲控制器启动发送操作, 按照 Buf— Out— p指示的序号, 向语音编码包提取单元提交语音会话 包, 每提交一个语音会话包, Buf— Out— p 自动增加一个单位, 其范 围为 0~ Max— Buf— Size-1 , 当 Buf— Out— p= Max— Buf— Size 时, 使 Buf— Out— p=0。
步骤 105, 接收端的语音编码包提取单元接收语音会话包, 去 封装并生成原始语音编码包。 语音编码包提取单元才艮据指定的语音 编码速率均匀输出原始语音编码包到应用层播放。 语音编码包提取 单元还可以在输出语音编码包缺失时根据序号插入一个或多个合 成的语音编码包或在输出语音编码包拥塞时删除一个或多个语音 编码包。 步骤 106, 发送端的语音包检测单元收到 NACK消息后, 判断当前 的带宽资源是否足够传输所有重传语音会话包及当前帧语音会话包 , 如 果是, 则发送所有重传语音会话包及当前帧语音会话包; 否则, 请求媒 体接入层, 执行资源重配置过程, 配置成功后, 再发送重传语音会话包 及新到达的语音会话包。
以上所述仅为本发明的较佳实施例而已, 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、 等同替换和改进等, 均应 包含在本发明的保护范围之内。

Claims

权利要求书
1、一种无线系统中传输语音的方法,其特征在于,该方法包括: 发送端对需发送的原始语音编码包标识用于表征播放顺序的 序号, 并对标识序号后的语音编码包进行信道编码构成语音会话 包;
发送端与接收端建立语音会话或语音数据混合会话, 为所述语 音会话或语音数据混合会话动态分配信道;
发送端根据预设的优先级对新到达语音会话包、 延迟语音会话 包、 重传语音会话包, 数据会话包及控制命令包进行发送;
接收端接收语音会话包并对所述语音会话包进行检测, 如果确 认语音会话包丢失, 向发送端发送包含丢失的语音会话包序号的
NACK包以指示发送端重传该语音会话包;
如果所述接收端为终端, 进一步包括: 将正确接收到的语音会 话包放入接收端的抖动緩冲控制器。
2、 根据权利要求 1 所述的方法, 其特征在于, 所述原始语音 编码包是如下语音编码包之一: G.723 , G.729, G.729A, G.729B , G.729D, G729E。
3、 根据权利要求 1 所述的方法, 其特征在于, 所述原始语音 编码包长度与语音会话包的长度比不小于 0.70。
4、 根据权利要求 1 所述的方法, 其特征在于, 所述发送端与 接收端建立语音会话或语音数据混合会话包括:
在会话建立请求消息中携带语音会话包的起始序号, 以使发送 端和接收端确定第一个语音会话包的序号。
5、 根据权利要求 1 所述的方法, 其特征在于, 所述发送端或 接收端为基站; 所述为所述语音会话或语音数据混合会话动态分配 信道包括:
基站根据当前帧需发送的语音会话包及数据会话包和当前信 道条件动态分配信道数目、 信道位置及语音会话包和数据会话包所 占用的每一信道对应的调制编码方式, 其中需发送的语音会话包包 括新到达的语音会话包、 被延迟的语音会话包或要重传的语音会话 包的其中之一或任意组合。
6、 根据权利要求 1 所述的方法, 其特征在于, 所述发送端或 接收端为基站; 所述为所述语音会话或语音数据混合会话动态分配 信道包括:
基站向终端发送带宽重配命令包以增加上行或下行带宽, 所述 增加的带宽在重传的语音会话包发送后自动取消, 无需基站向终端 发送控制命令包。
7、 根据权利要求 1 所述的方法, 其特征在于, 所述发送端或 接收端为基站; 所述为所述语音会话或语音数据混合会话动态分配 信道包括:
基站在检测到上行语音会话包丢失后向终端发送带宽重配命 令, 该命令不迟于包含丢失的语音会话包序号的 NACK 包到达终 端。
8、 根据权利要求 1 所述的方法, 其特征在于, 所述预设的优 先级由高到低依次为: 控制命令包、 重传语音会话包、 延迟语音会 话包、 新到达语音会话包、 数据会话包。
9、 根据权利要求 1所述的方法, 其特征在于, 进一步包括: 利用接收信噪比或检错码校验来确认语音会话包丢失。
10、 根据权利要求 1所述的方法, 其特征在于, 进一步包括: 将正确接收的语音会话包放入抖动緩冲控制器后, 抖动緩冲控 制器按所述序号的顺序勾速将正确接收到的语音会话包及插入的 合成语音编码包输出到所述无线系统的应用层。
11、 一种无线系统中传输语音的装置, 其特征在于, 该装置包 括:
包封装单元, 安装于基站和终端, 对需发送的原始语音编码包 标识用于表征播放顺序的序号, 并对标识序号后的语音编码包进行 信道编码构成语音会话包;
重发语音会话包产生器, 安装于基站和终端, 生成需要重传的 语音会话包;
信道分配单元, 安装于基站, 与终端交互语音会话建立请求消 息以建立语音会话或语音数据混合会话 , 动态分配信道;
发送包调度单元, 安装于基站和终端, 用于根据预设的优先级 对新到达语音会话包、 延迟语音会话包、 重传语音会话包、 数据会 话包及控制命令包进行发送;
语音包检测单元, 安装于基站和终端, 接收语音会话包并对所 述语音会话包进行检测, 如果确认语音会话包丢失, 将确认丢失的 语音会话包通知给 NACK包产生器;安装于终端的语音包检测单元 将正确收到的语音会话包送入抖动緩冲控制器; NACK包产生器, 安装于基站和终端, 生成包含丢失的语音会 话包序号的 NACK包,并向发送该丢失的语音会话包的发送端发送 NACK包以指示发送端重传该语音会话包。
12、 根据权利要求 11 所述的装置, 其特征在于, 所述包封装 单元中的原始语音编码包是如下语音编码包之一: G.723、 G.729、 G.729A, G.729B, G.729D, G729E。
13、 根据权利要求 11 所述的装置, 其特征在于, 所述包封装 单元中的原始语音编码包长度与语音会话包的长度比不小于 0.70。
14、 根据权利要求 11 所述的装置, 其特征在于, 所述信道分 配单元根据终端与基站交互的语音会话建立请求消息中携带的语 音包的起始序号, 来确定第一个语音会话包的序号。
15、 根据权利要求 11 所述的装置, 其特征在于, 所述基站中 的信道分配单元根据当前帧需发送的语音会话及数据会话包和当 前信道条件动态分配信道数目、 信道位置及语音会话包和数据会话 包所占用的每一信道对应的调制编码方式, 其中需发送的语音会话 包包括新到达的语音会话包、 被延迟的语音会话包或要重传的语音 ^舌包。
16、 根据权利要求 11 所述的装置, 其特征在于, 所述基站的 信道分配单元向终端发送带宽重配命令以增加上行或下行带宽, 所 述增加的带宽在重传的语音会话包发送后自动取消, 无需基站向终 端额外发送控制命令包。
17、 根据权利要求 11 所述的装置, 其特征在于, 所述基站的 信道分配单元在检测到上行语音会话包丢失后向终端发送带宽重 配命令,该命令不迟于包含丢失的语音会话包序号的 NACK包到达 终端。
18、 根据权利要求 11 所述的装置, 其特征在于, 所述预设的 优先级由高到低依次为: 控制命令包、 重传语音会话包、 延迟语音 会话包、 新到达语音会话包、 数据会话包。
19、 根据权利要求 11 所述的装置, 其特征在于, 所述语音包 检测单元, 利用接收信噪比或检错码校验来确认语音会话包丢失。
20、 根据权利要求 11 所述的装置, 其特征在于, 所述抖动緩 冲控制器, 安装于终端, 按所述序号的顺序匀速输出来自语音包检 测单元的正确接收到的语音会话包及插入的合成语音编码包到所 述无线系统的应用层。
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