WO2006081726A1 - Système de vidéoconférence multipoint et méthode de traitement de support de celui-ci - Google Patents

Système de vidéoconférence multipoint et méthode de traitement de support de celui-ci Download PDF

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Publication number
WO2006081726A1
WO2006081726A1 PCT/CN2005/002025 CN2005002025W WO2006081726A1 WO 2006081726 A1 WO2006081726 A1 WO 2006081726A1 CN 2005002025 W CN2005002025 W CN 2005002025W WO 2006081726 A1 WO2006081726 A1 WO 2006081726A1
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WIPO (PCT)
Prior art keywords
conference
video
terminal device
audio
chairperson
Prior art date
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PCT/CN2005/002025
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English (en)
French (fr)
Inventor
Yong Yang
Chun Yang
Yilin Cao
Kun Ding
Mingshi Huang
Xiaodong Gong
Original Assignee
Zte Corporation
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Publication date
Application filed by Zte Corporation filed Critical Zte Corporation
Priority to EP05814110A priority Critical patent/EP1850592A4/en
Publication of WO2006081726A1 publication Critical patent/WO2006081726A1/zh
Priority to US11/830,057 priority patent/US8767591B2/en

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/56Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • H04M3/567Multimedia conference systems
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N7/00Television systems
    • H04N7/14Systems for two-way working
    • H04N7/15Conference systems
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00Data switching networks
    • H04L12/02Details
    • H04L12/16Arrangements for providing special services to substations
    • H04L12/18Arrangements for providing special services to substations for broadcast or conference, e.g. multicast
    • H04L12/1813Arrangements for providing special services to substations for broadcast or conference, e.g. multicast for computer conferences, e.g. chat rooms
    • H04L12/1822Conducting the conference, e.g. admission, detection, selection or grouping of participants, correlating users to one or more conference sessions, prioritising transmission
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1101Session protocols
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066Session management
    • H04L65/1101Session protocols
    • H04L65/1104Session initiation protocol [SIP]
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/40Support for services or applications
    • H04L65/403Arrangements for multi-party communication, e.g. for conferences
    • H04L65/4046Arrangements for multi-party communication, e.g. for conferences with distributed floor control
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/47End-user applications
    • H04N21/478Supplemental services, e.g. displaying phone caller identification, shopping application
    • H04N21/4788Supplemental services, e.g. displaying phone caller identification, shopping application communicating with other users, e.g. chatting
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N7/00Television systems
    • H04N7/14Systems for two-way working
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04NPICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N7/00Television systems
    • H04N7/14Systems for two-way working
    • H04N7/15Conference systems
    • H04N7/152Multipoint control units therefor
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M2203/00Aspects of automatic or semi-automatic exchanges
    • H04M2203/50Aspects of automatic or semi-automatic exchanges related to audio conference
    • H04M2203/5054Meet-me conference, i.e. participants dial-in

Definitions

  • the present invention relates to video conferencing techniques, and more particularly to a multipoint video conferencing system and its media processing method. Background technique
  • the video conferencing system is a combination of multimedia computer technology and communication technology. Combined with multimedia technology and network communication technology, the video conferencing system provides an environment for people in different places to discuss and work together. It integrates computer interactivity, communication distribution, and the authenticity of TV. It has obvious superiority and has become a research hotspot in the computer field.
  • the early video conferencing system standard was the ITU-T H.320 standard developed by ITU-T (International Telecommunication Union Telecommunication Standardization Sector) in 1990. It is the basic standard for video conferencing systems based on circuit-switched bearers. With the continuous development of TCP/IP technology, ITU-T proposed the new generation video conferencing standard ITU-T H.323 in 1996, which is a video conferencing standard based on IP transport network, defined in the general network architecture. Above, regardless of the specific network, it greatly expands the scope of application.
  • the H.323 video conferencing system is rich and varied, each has its own strengths, but the interoperability of products from various manufacturers is not strong, which is inconvenient for the large-scale formation of the H.323 video conferencing system network.
  • a multi-point control unit MCU
  • the number of terminals that an existing video conferencing system can accommodate depends on the capabilities of the MCU. To expand the number of users, the configuration of the MCU must be increased. Therefore, high investment including other hardware devices will inevitably affect the wide application of the video conferencing system.
  • the SIP standard was proposed by the IETF (The Internet Engineering Task Force) organization in 1999. Its application goal is to realize real-time data, audio and video communication based on the Internet environment. Compared with H.323, the SIP (Session Initiation Protocol) protocol is relatively simple and free. Vendors can use existing network resources to expand multimedia communication services and construct systems that meet application requirements at a small cost. .
  • the SIP standard is strictly a signaling standard for real-time multimedia applications. It uses text-based coding to provide great flexibility, scalability, and cross-platform compatibility in a point-to-point application environment. .
  • SIP The protocol itself does not support multi-point conferencing functions as well as management and control functions. For the needs of multi-point communication, it is difficult to implement a simple SIP system.
  • the concept of a SIP video application server is proposed, which is composed of a focus, a media policy server, and a conference policies server.
  • the Center forms an overall management control and media transmission solution in a conference through appropriate media policies and conference policies, and is responsible for implementing the management of the conference and the distribution and transmission of the media.
  • the center functions like an MCU in an H.323-based video conferencing system.
  • the system still uses the traditional C/S architecture and requires a dedicated media control server to control and process the media.
  • the capabilities of the server will determine the number of users supported by the system. Summary of the invention
  • the object of the present invention is to provide a multi-point video conference system and a media processing method thereof, which can realize multi-point video conference without the MCU in the conventional video conference, and the networking is simple and convenient and reliable to implement.
  • the present invention provides a multipoint video conferencing system, including a softswitch device, an application server, and a terminal device, wherein the softswitch device is responsible for protocol adaptation and call processing, and is implemented as an external interface of the system.
  • the application server is used for control and accounting of business logic, characterized in that the terminal device that is a member of the conference includes the conference chairperson and the ordinary conference member, and any terminal device registered on the softswitch can be used as The conference chairperson and the conference chairperson must be SIP soft terminals with video conferencing service capabilities for media control.
  • the terminal device as the conference chairperson is a software terminal having multi-channel media control and processing capability; the terminal device is provided with a SIP module for performing a SIP session with the softswitch device and the application server; An audio processing module and a video processing module are provided.
  • the terminal that is the chairman of the conference is responsible for the scheduling of the conference, including initiating the conference, inviting members to join the conference, and ending the conference; the terminal device as the conference chairperson, while collecting local audio and video, The audio and video media packets from other conference members are received, and the received media packets are processed by the media processing unit inside the terminal device.
  • the audio processing module is responsible for collecting local audio and synthesizing, distributing and transmitting the received audio of other conference members; the video processing module is responsible for collecting local video and receiving other multiple channels. The video of the conference members is synthesized and transmitted.
  • the video processing module is responsible for segmentation and display of the video.
  • the call protocol includes but is not limited to SIP, MGCP or H.248, and the terminal device is a SIP terminal, an H.248 terminal or an MGCP terminal.
  • the present invention further provides a media processing method for a multipoint video conference system, in which, in an existing network, a multipoint video conference system is parsed into a conference chairperson and a plurality of ordinary conference members. Multiple sets of relatively independent SIP sessions, each group of sessions independently performing call processing and media negotiation work; after the primary and the called party are successfully connected in multiple sets of SIP calls, the terminal device as the conference chairperson completes the media processing, and respectively A plurality of conference members distribute and transmit the processed media stream; the media processing employs an end-to-end architecture.
  • the media processing comprises an audio processing step and a video processing step
  • the audio processing steps are as follows: the plurality of sets of SIP sessions independently perform audio media negotiation, and after the call starts, the conference chairperson simultaneously accepts from other multiple bits.
  • the audio of the conference members are decoded according to the corresponding coding format.
  • the local audio is collected, and the party and other conference members are processed according to different audio processing plans.
  • the audio processing plan is: synthesizing the audio of the first, second, and third members and playing the local And synthesizing the audio of the conference chairperson, the second and third members and encoding them according to the audio format negotiated between the conference chairperson and the first member, and then transmitting the audio to the first member; synthesizing the audio of the conference chairperson, the first member and the third member And transmitting to the second member according to the audio format code negotiated between the conference chairperson and the second member; synthesizing the audio of the conference chairperson, the first and second members, and encoding according to the audio format negotiated between the conference chairperson and the third member Send to the third member.
  • the conference chairperson negotiates the video in the first format; after the call is established, the conference chairperson will receive the other conference members.
  • the multi-channel video is decoded and sampled into the second format, and the locally collected video is also sampled into the second format, and then synthesized into a complete frame of the first format video, which is respectively sent to other conference members;
  • the member decodes the received video and then splits the video of other meeting members and displays them in separate screens.
  • the conference member includes the conference chairperson and the first conference member
  • the second conference member when the conference member includes the conference chairperson and the first conference member, the second conference member, the first format is a QCIF format, and the second format is a quarter QCIF format.
  • the multi-point video conference system and the media processing method thereof are simple and convenient, and the system has good practicability and scalability.
  • the call processing in the present invention is implemented based on the SIP protocol, and is simple and free relative to H.323.
  • the control and processing of the media is performed by the client as the conference chairperson, and thus, the system that satisfies the application can be constructed in the existing NGN network, so that the operator can easily realize the large-scale service. Promotion and application;
  • the control of the media in the present invention is implemented by means of a soft terminal as the chairman of the conference, so that the end users registered on the softswitch can initiate or participate in a multi-party multi-point video conference, while in the traditional H.323-based standard.
  • the number of terminals that can be accommodated is limited by the capabilities of the MCU; the soft terminal in the present invention has the capability of simultaneously performing multiple SIP sessions, and also has media such as media synthesis, distribution, transmission, and segmentation. control function.
  • the terminal devices used by the conference members are not limited to ordinary SIP software terminals or hardware terminals, but also applicable to H.323 terminals;
  • the business logic is controlled by an application server, which is based on the Parlay/OSA architecture. Due to the openness of the Parlay/OSA specification, the implementation of the business logic is more flexible and convenient, and can meet the needs of different customer groups. At the same time, the application server has a reliable and flexible charging strategy, which can be tailored according to the needs of the service and the operator. Billing plan. DRAWINGS
  • FIG. 1 is a structural diagram of a multipoint video conference system of the present invention
  • FIG. 2 is a call flow diagram in the method of the present invention.
  • Figure 3 is a flow chart of media transmission in the method of the present invention.
  • Figure 4 is a flow chart of audio processing in the method of the present invention.
  • FIG. 5 is a flow diagram of video processing in the method of the present invention. detailed description
  • FIG. 1 is a system structural diagram of a multipoint video conference system according to the present invention.
  • the system of the present invention includes: a softswitch device 1, an application server 2, and a terminal device 3.
  • the softswitch device 1 mainly performs the following functions:
  • Protocol adaptation and call processing which implements adaptation of protocols such as SIP, H.248, and MGCP (Media Gateway Control Proctol), and simultaneously receives and forwards call signaling.
  • the application server 2 is developed based on the Parlay/OSA specification, and the SIP call proxy service module is integrated internally to implement the following functions:
  • End-user authentication and authentication Only a terminal user authenticated and authenticated by the application server 2 can initiate a multi-party multi-point video conference. Otherwise, it can only be invited to join the conference as a conference member.
  • Control and billing of business logic accepting the conference request (XML language) from the conference chairperson, and parsing it into multiple sets of SIP sessions, and then sending SIP messages to the softswitch device 1 through the internal SIP call proxy service module, After the call is established, it is responsible for real-time charging of the conference.
  • conference request XML language
  • SIP call proxy service module After the call is established, it is responsible for real-time charging of the conference.
  • the terminal device 3 includes: a SIP soft terminal or a normal SIP hard terminal, an H.248 terminal, an MGCP terminal or an H.323 terminal (the H.323 terminal is interconnected with the softswitch device 1 through the H.323 gateway).
  • the ordinary SIP hard terminal and the H.323 terminal can only be invited to join the conference as the ordinary conference member.
  • the SIP soft terminal can initiate a conference as the conference chairperson or be invited to join the conference as a member of the ordinary conference.
  • the standard SIP protocol is used for interaction.
  • a SIP soft terminal initiates a conference request, it encapsulates the conference request described in the XML language into the SIP MESSAGE source language and sends it to the application server.
  • the embodiment is described by taking a four-party conference as an example.
  • a single conference in the system can support four-party multi-point video and audio communication, and the number of conferences is unlimited. All terminals registered on the softswitch device can participate in a four-party multipoint video conference initiated by the SIP soft terminal.
  • the client A as the conference chairperson sends a conference initiation request in XML language to the application server, requesting to hold a four-party video conference of A and users B, C, and D;
  • the application server parses the received XML message, and uses the conference chair A as the calling party to send a SIP request to the softswitch device, and hopes to establish a call between the terminal device A and 8, C, and D respectively;
  • the softswitch device controls and processes the call signaling, and establishes calls between eight and 8, A and C, and A and D, respectively; 4) When 8, C or D hangs up, A can continue to invite other clients to participate in the conference through the application server; when A hangs up, notify the application server to end the entire conference.
  • RTP Real-time Transport Protocol
  • UDP User Datagram Protocol
  • RTP Real-time Transport Protocol
  • UDP User Datagram Protocol
  • RTP is defined to work in the case of one-to-one or one-to-many transmissions, the purpose of which is to provide time information and achieve stream synchronization.
  • RTP When the application starts an RTP session, it will use two ports: one for RTP and one for RTCP (Real-time Transport Control Protocol).
  • RTP itself does not provide a reliable transport mechanism for delivering packets in sequence, nor does it provide flow control or congestion control. It relies on RTCP to provide these services.
  • the synthesis of four channels of audio and the processing of video are done by client A, who is the chairman of the conference.
  • the encoding and decoding of audio and video are all implemented by software.
  • the audio codec supports G.711, G.729 and G.723, the video uses the H.263 standard, QCIF format.
  • Terminal devices B, C, and D which are members of the conference, will open two pairs of RTP and RTCP ports respectively after receiving the calls with A, receive audio and video media packets from A, and simultaneously The specified port sends the locally collected audio media package and video media package.
  • this process is identical to normal point-to-point videophone.
  • the audio codec in the present invention supports G.711, G.729, and G.723, and the audio processing process is as shown in FIG. 4, after the SIP sessions of eight, 8, C, and D are established:
  • Conference chair A collects local audio and receives three channels of audio from conference members B, C and D and decodes them separately;
  • Conference Chair A will decode the audio mix of B, C and D and play it through the speaker; mix the local audio with the audio of C and D, compress and encode it and send it to B; mix the local audio with the audio of B and D. , after compression encoding, send to C; mix local audio with audio of B and C, press After encoding, send it to D;
  • Ordinary meeting members B, C, and D receive the mix from the chairperson A of the meeting, and then play it out through the speaker after decoding.
  • the video codec in the present invention adopts the H.263 standard, and the maximum bandwidth is limited to 384k bps, and the video processing process is as shown in FIG. 5:
  • Conference Chair A collects local video (QCIF format) and sends back the local display, and performs sampling processing at the same time. The size is sampled by QCIF as a quarter QCIF;
  • the conference chairperson A receives the three video packets from conference members B, C and D, decodes them separately, and sends them to the local video display window for display;
  • QCIF sampling is one quarter of QCIF
  • the conference chairperson A compresses and encodes the stitched QCIF images and sends them to conference members B, C and D respectively;
  • Conference members B, C, and D receive the video media package from conference chair A, decode, split and get the other three parties' video and display them, and simultaneously send the locally collected video to the local video display window for display. It should be noted that for a SIP hard terminal or an H.323 terminal that is a conference member, only the received video can be directly sent to the video display window display. Currently, the image of each conference member cannot be displayed on the split screen.
  • the networking of the invention is simple and convenient, and the system has good practicability and scalability.
  • call processing is implemented based on the SIP protocol, which is simple and free compared to H.323.
  • the control and processing of the media is performed by the client as the conference chairman, and thus, the system that satisfies the application can be constructed in the existing NGN network, so that the operator can realize the large-scale service very conveniently. Promotion and application;
  • the control of the media in the present invention is realized by relying on a soft terminal as the chairman of the conference, and thus in soft handing
  • the registered end user can initiate or participate in a multi-party multi-point video conference;
  • the soft terminal in the present invention has the capability of simultaneously performing multiple SIP sessions, and also has media control such as media synthesis, distribution, transmission, and segmentation.
  • media control such as media synthesis, distribution, transmission, and segmentation.
  • the terminal device used by the conference members is not limited to an ordinary SIP software terminal or a hardware terminal, but is also applicable to an H.323 terminal;
  • the business logic is controlled by an application server, which is based on the Parlay/OS A architecture. Due to the openness of the Parlay/OS A specification, the implementation of the business logic is more flexible and convenient, and can meet the needs of different customer groups. At the same time, the application server has a reliable and flexible charging strategy, which can be tailored according to the needs of the service and the operator. Billing plan.
  • the core device softswitch device of the next generation network NGN is a call processing device, and the application server developed based on the Pariay/OSA specification controls the business logic, parses the video conference into multiple sets of point-to-point SIP sessions, and organically ingeniously Combined, in the media processing, the traditional C / S (terminal / server) mode is abandoned, the client-to-client (end-to-end) architecture is adopted, and the media processing is handled by the terminal device as the chairman of the conference.
  • the land expands the support capabilities of video conferencing systems. This service can be carried out in the existing NGN network without adding other equipment. For the end user, the configuration is simple and easy to use.

Description

一种多点视频会议系统及其媒体处理方法 技术领域
本发明涉及视频会议技术特别是一种多点视频会议系统及其媒体处理方 法。 背景技术
视频会议系统是多媒体计算机技术与通信技术相结合的产物。结合多媒体 技术和网络通信技术,视频会议系统为身处异地的人们提供了一个相互讨论问 题并协同工作的环境。它集计算机的交互性、通信的分布性, 及电视的真实性 为一体, 具有明显的优越性, 因而成为当今计算机领域的一个研究热点。
早期的视频会议系统标准为 ITU-T (国际电信联盟电信标准化部门) 于 1990年制订的 ITU-T H.320标准,它是针对基于电路交换承载的视频会议系统 的基础标准。随着 TCP/IP技术的不断发展, ITU-T于 1996年提出了新一代视 频会议标准 ITU-T H.323,该标准是基于 IP传输网络实现承载的视频会议标准, 定义在通用网络体系结构之上, 与具体网络无关, 极大拓宽了应用范围。
从产品现状来看, 尽管 H.323视频会议系统丰富多样, 各有所长, 但是各 个厂家的产品互连互通能力不强,这对于大规模组建 H.323视频会议系统网络 造成了不便。 在现有的以 H.323 为标准的视频会议系统中, 由多点控制单元 MCU (Multi-point Control Unit) 作为核心的媒体控制器, 负责各个终端之间 的语音和图像数据的传递。 现有视频会议系统所能容纳的终端个数将取决于 MCU的能力, 要扩展用户数目, 必须增加 MCU的配置, 因而包括其它硬件 设备在内的高昂投资也势必影响视频会议系统的广泛应用。
SIP标准是 IETF (The Internet Engineering Task Force,互联网工程任务组) 组织在 1999年提出的,其应用目标是基于 Internet环境,实现数据、音视频实 时通讯。 SIP ( Session Initiation Protocol, 会话初始化协议)协议相对于 H.323 而言, 相对简单、 自由, 厂商可以利用现有的网络资源扩展多媒体通讯业务, 以较小的成本就可以构造满足应用需求的系统。 SIP标准严格意义上讲是一个 实现实时多媒体应用的信令标准,它采用基于文本的编码方式,在点到点的应 用环境中, 具有极大的灵活性、 扩充性以及跨平台使用的兼容性。 但是, SIP 协议自身不支持多点的会议功能以及管理和控制功能, 对于多点通讯的需求, 应用单纯的 SIP系统难以实现。
因而,在现有的基于 SIP的软件视频会议解决方案中,提出了 SIP视频应 用服务器的概念, 该服务器由中心(Focus) 、 媒体策略(Media Policy)服务 器、会议策略(Conference Policies)服务器组成。 中心通过适当的媒体策略和 会议策略形成一次会议中的整体管理控制和媒体传输方案,负责实施对会议的 管理及媒体的分发和传输。在一定程度上讲, 中心的功能类似于基于 H.323的 视频会议系统中的 MCU。 该系统仍然沿用传统的 C/S架构, 需要专门的媒体 控制服务器来控制和处理媒体。 服务器的能力将决定系统所支持的用户数。 发明内容
本发明的目的在于提供一种多点视频会议系统及其媒体处理方法,无需传 统视频会议中的 MCU即可实现多点视频会议, 同时组网简单, 实施起来方便 可靠。
为了实现上述目的,本发明提供了一种多点视频会议系统,包括软交换设 备、应用服务器和终端设备,其中,软交换设备用于负责协议适配和呼叫处理, 同时作为系统的对外接口完成和其它系统的互连互通,应用服务器用于业务逻 辑的控制及计费,其特征在于:作为会议成员的终端设备包括会议主席和普通 会议成员,任何注册在软交换上的终端设备均可作为会议成员,会议主席必须 是具有视频会议业务能力的 SIP软终端, 用于实现媒体的控制。
上述的系统,其中,所述作为会议主席的终端设备是具有多路媒体控制和 处理能力的软件终端; 终端设备设置有 SIP模块,用于与软交换设备和应用服 务器进行 SIP会话; 终端设备还设置有音频处理模块和视频处理模块。
上述的系统, 其中, 作为会议主席的终端设^ "负责会议的调度工作, 包括 发起会议, 邀请成员加入会议, 结束会议; 作为会议主席的终端设备, 在采集 本地音频和视频的同时, 还将接收来自于其他会议成员的音频和视频媒体包, 由终端设备内部的媒体处理单元处理接收到的媒体包。
上述的系统, 其中, 作为会议主席的终端设备中, 音频处理模块负责采集 本地音频并和接收到的其它会议成员的音频进行合成,分发与传输;视频处理 模块负责采集本地视频、 接收其它多路会议成员的视频并进行合成与传输。 上述的系统, 其中, 作为普通会议成员的终端设备中,所述视频处理模块 负责视频的分割与显示。
上述的系统, 其中, 呼叫协议包括但不限于 SIP、 MGCP或 H.248, 终端 设备为 SIP终端、 H.248终端或 MGCP终端。
为了更好的实现上述目的,本发明还提供了一种多点视频会议系统的媒体 处理方法, 其中, 在现有网络中, 将多点视频会议系统解析为会议主席和多个 普通会议成员之间的多组相对独立的 SIP会话,每组会话独立地进行呼叫处理 和媒体协商工作;在多组 SIP呼叫中主被叫连接成功之后, 由作为会议主席的 终端设备完成媒体处理,并分别向多个会议成员分发和传输经过处理之后的媒 体流; 所述媒体处理采用端到端的架构。
上述的方法, 其中,所述媒体处理包括音频处理步骤和视频处理步骤, 其 中音频处理步骤如下: 多组 SIP会话独立进行音频媒体的协商,在通话开始之 后,会议主席同时接受来自于其它多位会议成员的音频,分别按照相应的编码 格式进行解码; 同时采集本地音频,对本方和其它会议成员分别按照不同的音 频处理计划进行处理。
上述的方法, 其中, 当会议成员包括会议主席和普通会议成员第一、第二 和第三成员时, 所述音频处理计划为: 将第一、第二和第三成员的音频合成后 本地播放;将会议主席、第二和第三成员的音频合成并按照会议主席与第一成 员之间协商的音频格式编码之后发送给第一成员;将会议主席、第一和第三成 员的音频合成并按照会议主席与第二成员之间协商的音频格式编码之后发送 给第二成员;将会议主席、第一和第二成员的音频合成并按照会议主席与第三 成员之间协商的音频格式编码之后发送给第三成员。
上述的方法, 其中, 所述视频处理步骤如下: 在多组 SIP会话的媒体协商 过程中,会议主席均以第一格式协商视频; 在通话建立之后,会议主席将接收 到的来自于其他会议成员的多路视频解码并抽样成第二格式,同时将本地采集 到的视频也抽样为第二格式,然后再合成为一帧完整的第一格式的视频,分别 发送给其它多个会议成员;会议成员解码接收到的视频,然后分割出其它会议 成员的视频并分屏显示。
上述的方法, 当会议成员包括会议主席和普通会议成员第一、第二和第三 成员时, 所述第一格式为 QCIF格式, 所述第二格式为四分之一 QCIF格式。 本发明的多点视频会议系统及其媒体处理方法组网简单方便,系统具有良 好的实用性和可扩展性。本发明中呼叫处理是基于 SIP协议来实现的,相对于 H.323要简单、 自由。 同时, 在本发明中, 媒体的控制和处理由作为会议主席 的客户端来完成, 因而, 在现有的 NGN网络中就可以构造满足应用的系统, 使得运营商可以非常方便实现大规模的业务推广和应用;
本发明中媒体的控制是依靠作为会议主席的软终端来实现的,因而在软交 换上注册的终端用户均可发起或参与一个多方的多点视频会议,而在传统的基 于 H.323标准的视频会议系统中,其所能容纳的终端个数受限于 MCU的能力; 本发明中的软终端具有同时进行多路 SIP会话的能力,同时还具有媒体的 合成、分发、传输和分割等媒体控制功能。 当然, 考虑到软交换设备的强大的 协议适配能力,会议成员所使用的终端设备,不仅局限于普通的 SIP软件终端 或硬件终端, 还适用于 H.323终端;
本发明中业务逻辑是由应用服务器来控制的, 而该应用服务器是建立 在 Parlay/OSA架构基础之上的。 由于 Parlay/OSA规范的开放性,使得业务逻 辑的实现更加灵活方便, 可以满足不同客户群的需要; 同时,应用服务器具有 可靠和灵活的计费策略, 可以根据业务和运营商的需要制订相应的计费方案。 附图说明
图 1是本发明的多点视频会议系统的结构图;
图 2是本发明方法中的呼叫流程图;
图 3是本发明方法中的媒体传输流程图;
图 4是本发明方法中的音频处理流程图;
图 5是本发明方法中的视频处理流程图。 具体实施方式
图 1所示为本发明的多点视频会议系统的系统结构图,如图 1所示,本发 明系统包括: 软交换设备 1, 应用服务器 2和终端设备 3。
其中软交换设备 1主要完成以下功能:
接受终端设备 3的登记和注册,其中 SIP软终端或硬终端直接在软交换设 备 1上注册, H.323终端通过 H.323网关到软交换设备上注册; 协议适配和呼叫处理,实现 SIP、 H.248和 MGCP (Media Gateway Control Proctol, 媒体网关控制协议)等协议的适配, 同时接收和转发呼叫信令。
应用服务器 2基于 Parlay/OSA规范开发, 同时内部集成有 SIP的呼叫代 理服务模块, 用于实现以下功能:
终端用户的认证和鉴权, 只有通过应用服务器 2认证和鉴权的终端用户, 才能发起一个多方的多点视频会议,否则只能以会议成员的身份被邀请加入会 议;
业务逻辑的控制及计费, 接受来自于会议主席的会议申请(XML语言), 并将其解析为多组 SIP会话, 然后通过内部的 SIP呼叫代理服务模块发送 SIP 消息给软交换设备 1, 同时, 在通话建立之后, 负责对会议进行实时计费。
终端设备 3包括: SIP软终端或普通的 SIP硬终端、 H.248终端、 MGCP 终端或 H.323终端(H.323终端通过 H.323网关与软交换设备 1互联) 。 其中 普通的 SIP硬终端及 H.323终端只能以普通会议成员的身份被邀请加入会议, 而 SIP软终端既可以会议主席的身份发起一个会议,也可以作为普通会议成员 被邀请加入会议。
软交换设备 1和应用服务器 2之间,软交换设备 1和 SIP终端之间以及应 用服务器 2和通过它鉴权与认证的 SIP软终端之间,均按照标准的 SIP协议进 行交互。 其中, 当一个 SIP软终端发起一个会议请求时, 它会将以 XML语言 描述的会议请求封装在 SIP MESSAGE源语之中, 发送给应用服务器。
实施例以四方会议为例进行说明。系统中单个会议可支持四方的多点视频 和音频通讯, 会议个数不限, 在软交换设备上注册的所有终端均可参与由 SIP 软终端发起的一个四方的多点视频会议。
如图 2所示,对于四方的多点视频会议,其呼叫流程被解析为 3组相对独 立的 SIP会话, 过程如下:
1)作为会议主席的客户端 A向应用服务器发送以 XML语言描述的会议发 起请求, 请求召开 A和用户 B、 C、 D的四方视频会议;
2)应用服务器解析接收到的 XML消息, 以会议主席 A为主叫, 向软交换 设备发送 SIP请求, 希望分别建立终端设备 A与8、 C、 D间的呼叫;
3)软交换设备控制并处理呼叫信令, 分别建立八与8、 A与 C及 A与 D 之间的呼叫; 4)当8、 C或 D挂机时, A可以继续通过应用服务器邀请其它客户端参加 会议; 当 A挂机时, 通知应用服务器结束整个会议。
其中, 媒体的传输采用基于 UDP (User Datagram Protocol, 用户数据报协 议) 的 RTP (Real-time Transport Protocol, 实时传输协议)来完成, RTP是用 于 Internet上针对多媒体数据流的一种传输协议。 RTP被定义为在一对一或一 对多的传输情况下工作,其目的是提供时间信息和实现流同步。当应用程序开 始一个 RTP会话时将使用两个端口: 一个给 RTP, 一个给 RTCP (Real-time Transport Control Protocol, 实时传输控制协议)。 RTP本身并不能为按顺序传 送数据包提供可靠的传送机制, 也不提供流量控制或拥塞控制, 它依靠 RTCP 提供这些服务。
如图 3所示, 四路音频的合成和视频的处理由作为会议主席的客户端 A 来完成, 音频和视频的编解码全部采用软件来实现, 音频编解码支持 G.711、 G.729和 G.723, 视频采用 H.263标准, QCIF格式。
1)对于会议主席 A, 当它与 B、 C和 D的呼叫建立之后, 会同时打开 3组 媒体通道,每一组同时打开 2对 RTP和 RTCP端口,分别接收来自 B、 C和 D 的音频媒体包与视频媒体包。 同时, 八向3、 C、 D上指定的端口发送处理过 后的音频媒体包与视频媒体包;
2)作为会议成员的终端设备 B、 C和 D,在它们与 A的呼叫分别建立之后, 将各自打开 2对 RTP和 RTCP端口, 接收来自于 A的音频和视频媒体包, 同 时, 向 A上指定的端口发送本地采集到的音频媒体包与视频媒体包。 对终端 设备 B、 C和 D而言, 此过程与普通的点对点可视电话并无二致。
本发明中的音频编解码支持 G.711、 G.729和 G.723,音频处理过程如图 4 所示, 在八与8、 C和 D的 SIP会话建立之后:
1)普通会议成员 B、 C和 D分别采集本地音频,在压缩编码之后发送给会 议主席 A;
2)会议主席 A采集本地音频, 同时接收到来自于会议成员 B、 C和 D的 三路音频并分别进行解码;
3)会议主席 A将解码得到 B、 C和 D的音频混合, 通过扬声器播放出来; 将本地音频与 C、 D的音频混合, 压缩编码之后发送给 B; 将本地音频与 B、 D的音频混合, 压缩编码之后发送给 C; 将本地音频与 B、 C的音频混合, 压 缩编码之后发送给 D;
4)普通会议成员 B、 C和 D分别接收来自于会议主席 A的混音, 解码之 后通过扬声器播放出来。
本发明中的视频编解码采用 H.263标准, 最大带宽限定在 384k bps,视频 处理过程如图 5所示:
1)会议主席 A采集本地视频(QCIF格式) 并回送本地显示, 同时进行抽 样处理, 尺寸由 QCIF抽样为四分之一 QCIF;
2)会议主席 A接收到来自于会议成员 B、 C和 D的三路视频包后, 分别 进行解码, 并送给本地视频显示窗口显示;
3)会议主席 A对解码后得到的各帧图像进行抽样, 每帧图像的尺寸由
QCIF抽样为四分之一 QCIF;
4)会议主席 A将四帧四分之一 QCIF的图像拼接为一帧 QCIF的图像;
5)会议主席 A将拼接得到的 QCIF的图像进行压缩和编码,分别发送给会 议成员 B、 C和 D;
6)会议成员 B、 C和 D接收到来自会议主席 A的视频媒体包, 进行解码, 分割得到其它三方的视频并进行显示, 同时,各自将本地采集到的视频送到本 地视频显示窗口显示。需要指出的是,对于作为会议成员的 SIP硬终端或 H.323 终端,只能将接收到的视频直接送到视频显示窗口显示, 目前无法实现分屏显 示各个会议成员的图像。
当然,本发明还可有其他多种实施例,在不背离本发明精神及其实质的情 况下,熟悉本领域的技术人员当可根据本发明作出各种相应的改变和变形,但 这些相应的改变和变形都应属于本发明所附的权利要求的保护范围。 工业应用性
本发明组网简单方便,系统具有良好的实用性和可扩展性。本发明中呼叫 处理是基于 SIP协议来实现的, 相对于 H.323要简单、 自由。 同时, 在本发明 中,媒体的控制和处理由作为会议主席的客户端来完成,因而,在现有的 NGN 网络中就可以构造满足应用的系统,使得运营商可以非常方便实现大规模的业 务推广和应用;
本发明中媒体的控制是依靠作为会议主席的软终端来实现的,因而在软交 换上注册的终端用户均可发起或参与一个多方的多点视频会议; 本发明中的软终端具有同时进行多路 SIP会话的能力,同时还具有媒体的 合成、分发、传输和分割等媒体控制功能。 当然, 考虑到软交换设备的强大的 协议适配能力,会议成员所使用的终端设备, 不仅局限于普通的 SIP软件终端 或硬件终端, 还适用于 H.323终端;
本发明中业务逻辑是由应用服务器来控制的, 而该应用服务器是建立 在 Parlay/OS A架构基础之上的。 由于 Parlay/OS A规范的开放性,使得业务逻 辑的实现更加灵活方便, 可以满足不同客户群的需要; 同时, 应用服务器具有 可靠和灵活的计费策略, 可以根据业务和运营商的需要制订相应的计费方案。
本发明以下一代网络 NGN的核心设备软交换 设备为呼叫处理设备, 以 基于 Pariay/OSA规范开发的应用服务器来控制业务逻辑, 将视频会议解析为 多组点对点的 SIP会话, 并将其有机地巧妙地结合起来, 在媒体处理上, 摒弃 了传统的 C/S (终端 /服务器)模式, 采用 Client-to-Client (端到端) 的架构, 媒体处理由作为会议主席的终端设备来处理,极大地拓展了视频会议系统的支 持能力。 在现有的 NGN网络中即可开展该项业务, 无需增加其它设备。 对于 终端用户而言, 配置简单, 使用方便。

Claims

权利要求书
1、 一种多点视频会议系统, 用于现有网络, 包括软交换设备、 应用服务 器和终端设备, 其中:
所述应用服务器用于根据作为会议主席的终端设备的多点视频会议请求 通知所述软交换设备分别建立作为会议主席的终端设备和普通会议成员之间 的终端设备的呼叫;
所述软交换设备用于协议适配和呼叫处理,分别建立作为会议主席的终端 设备和作为普通会议成员的终端设备之间的独立的呼叫;
所述终端设备包括作为会议主席的终端设备和作为普通会议成员的终端 设备, 作为普通会议成员的终端设备为任何注册在软交换设备上的终端设备; 作为会议主席的终端设备为具有视频会议业务能力的 SIP软终端,用于实现媒 体的控制,并分别向作为普通会议成员的终端设备分发和传输经过处理之后的 媒体流。
2、 根据权利要求 1所述的系统, 其特征在于, 所述作为会议主席的终端 设备是具有多路媒体控制和处理能力的软件终端; 终端设备设置有 SIP模块, 用于与软交换设备和应用服务器进行 SIP会话;终端设备还设置有音频处理模 块和视频处理模块。
3、 根据权利要求 2所述的系统, 其特征在于, 作为会议主席的终端设备 用于会议的调度工作, 包括发起会议, 邀请成员加入会议, 结束会议; 作为会 议主席的终端设备,用于在采集本地音频和视频的同时接收来自于其他会议成 员的音频和视频媒体包, 由终端设备内部的媒体处理单元处理接收到的媒体 包。
4、 根据权利要求 3所述的系统, 其特征在于, 作为会议主席的终端设备 中,音频处理模块用于采集本地音频,并和接收到的其它会议成员的音频进行 合成, 分发与传输; 视频处理模块用于采集本地视频、接收其它多路会议成员 的视频并进行合成与传输。
5、 根据权利要求 2所述的系统, 其特征在于, 作为普通会议成员的终端 设备中, 所述视频处理模块负责视频的分割与显示。
6、根据权利要求 1所述的系统,其特征在于,呼叫协议包括但不限于 SIP、 MGCP或 H.248, 终端设备为 SIP终端、 H.248终端或 MGCP终端。
7、一种多点视频会议系统的媒体处理方法, 其特征在于, 在现有网络中, 将多点视频会议请求解析为会议主席和多个普通会议成员之间的多组相对独 立的 SIP会话,每组会话独立地进行呼叫处理和媒体协商工作; 在多组 SIP呼 叫中主被叫连接成功之后, 由作为会议主席的终端设备完成媒体处理,并分别 向多个会议成员分发和传输经过处理之后的媒体流;所 k媒体处理采用端到端 的架构。
8、 根据权利要求 7所述的方法, 其特征在于, 所述媒体处理包括音频处 理步骤和视频处理步骤,其中音频处理步骤如下: 多组 SIP会话独立进行音频 媒体的协商,在通话开始之后,会议主席同时接受来自于其它多位会议成员的 音频, 分别按照相应的编码格式进行解码; 同时采集本地音频, 对本方和其它 会议成员分别按照不同的音频处理计划进行处理。
9、 根据权利要求 8所述的方法, 其特征在于, 当会议成员包括会议主席 和普通会议成员第一、第二和第三成员时, 所述音频处理计划为: 将第一、第 二和第三成员的音频合成后本地播放;将会议主席、第二和第三成员的音频合 成并按照会议主席与第一成员之间协商的音频格式编码之后发送给第一成员; 将会议主席、第一和第三成员的音频合成并按照会议主席与第二成员之间协商 的音频格式编码之后发送给第二成员;将会议主席、第一和第二成员的音频合 成并按照会议主席与第三成员之间协商的音频格式编码之后发送给第三成员。
10、 根据权利要求 8所述的方法, 其特征在于, 所述视频处理步骤如下: 在多组 SIP会话的媒体协商过程中,会议主席均以第一格式协商视频;在通话 建立之后,会议主席将接收到的来自于其他会议成员的多路视频解码并抽样成 第二格式, 同时将本地采集到的视频也抽样为第二格式,然后再合成为一帧完 整的第一格式的视频,分别发送给其它多个会议成员;会议成员解码接收到的 视频, 然后分割出其它会议成员的视频并分屏显示。
11、 根据权利要求 10所述的方法, 其特征在于, 当会议成员包括会议主 席和普通会议成员第一、 第二和第三成员时, 所述第一格式为 QCIF格式, 所 述第二格式为四分之一 QCIF格式。
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