CN101695058A - 在数据网络通话系统中提供通信服务的方法 - Google Patents

在数据网络通话系统中提供通信服务的方法 Download PDF

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CN101695058A
CN101695058A CN200910180890A CN200910180890A CN101695058A CN 101695058 A CN101695058 A CN 101695058A CN 200910180890 A CN200910180890 A CN 200910180890A CN 200910180890 A CN200910180890 A CN 200910180890A CN 101695058 A CN101695058 A CN 101695058A
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message
user agent
server
call
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昆丹·辛
海宁·舒尔泽内
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Columbia University of New York
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    • H04M7/1205—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
    • H04M7/1295—Details of dual tone multiple frequency signalling

Abstract

本发明总的来说涉及互联网和内联网通话领域，尤其涉及一种在数据网络通话系统中提供通信服务的方法。一种在数据网络通话系统中提供通信服务的方法，包括：接收从第一用户代理到第二用户代理的呼叫请求；对第二用户代理产生一呼叫请求并对通信服务器产生与第二用户代理相关的的呼叫请求；在为与媒体服务器进行消息交换而提供一消息条件时，在第一用户代理和媒体服务器之间建立一个链接；若没有消息条件发生，终止向通信服务器的呼叫请求。

Description

在数据网络通话系统中提供通信服务的方法
本申请是申请号为01813928.0的发明专利申请的分案申请。
原案申请日：2001年8月13日
原案发明名称：在互联网/内联网通话中统一通信的系统
原案申请号：01813928.0，国际申请号：PCT/US01/41694
发明领域
本发明总的来说涉及互联网和内联网通话领域，尤其涉及一种在数据网络通话系统中提供通信服务的方法。
背景技术
互联网已经发展成一种在商业、技术和教育领域中的数百万用户的基本通信工具。在这方面，互联网正在成长的用途涉及互联网通话，所述互联网通话具有许多优点优于由各个信号网络控制的常规线路交换网络。
多数现代通话系统的重要特征是声音通信。声音通信的范围是统一通信，其中在共同的平台上以各种媒体形式对通信服务进行访问。例如，在公共系统内，作为通话网络一部分的转换声音邮件、电子邮件、视频通信、即时通信服务及其类似服务的能力为用户提供了单一平台，从而可方便地利用这些服务。
声音邮件服务通常由局部私人交换分机(PBX)或局部交换载波提供。这类通用声音邮件系统是典型的封闭体系结构。结果，难以执行简单操作，如在局部PBX外转寄声音邮件，过滤或排序信息。因此，希望利用内部简单数据交换和没有局部通话交换的开放体系结构。
会话启动协议(SIP)作为用于互联网通话的标准发送协议正在得到普及。随着这种普及的发展，希望提供一种系统体系结构和基于SIP系统提供统一通话服务的方法。此外，在互联网上已广泛采用实时流协议(RTSP)作为标准传输协议，用于多媒体服务，如视频、音频和混合媒体文件。利用SIP作为发送协议及RTSP作为消息存贮和传输的统一通信系统中可具有许多优于已知的通信系统的优点。
发明内容
根据本发明，提供一种能够进行统一通信服务网络通话系统。系统总体上包括数据网络，至少一个可操作地连接到数据网络上的用户代理和可操作地连接到数据网络上的发送信号服务器。用户代理是通话端点，如独立的互联网通话应用或具有适当通话软件的个人计算机。提供一通信服务器可操作地连接到数据网络上并响应发送信号服务器。该系统还包括媒体服务器，其可操作地连接到网络上并包括用来存贮消息文件的计算机数据存贮媒体。媒体服务器响应通信服务器，并根据消息发生条件直接访问呼叫用户，以便接收媒体服务器上的录制的消息文件或存贮用于被叫用户下一步检索的消息文件。
更适宜的是，发送信号服务器包括用于注册用户代理的网络地址数据库，如通话启动协议(SIP)地址。根据从呼叫用户接收到的呼叫请求，发送信号服务器产生一个对存贮在数据库中的被叫用户代理地址的呼叫请求，还产生一个对通信服务器的网络地址的呼叫请求。如果对用户代理的呼叫请求被接收到，发送信号服务器可取消向通信服务器的呼叫请求。如果探测到消息条件，发送信号服务器还可取消向被叫用户代理的呼叫请求。一个这样的消息条件是呼叫请求启动后的一段预定时间。
在一个实施例中，当探测到消息条件后，可对被叫用户代理地址提供邀请请求，同时将消息提供到媒体服务器上。利用接收的邀请请求，被叫用户代理可从媒体服务器上恢复呼叫。
当探测到消息条件时，媒体服务器对呼叫用户提供一个存贮的的问候消息如提示留下消息。媒体服务器优选支持流媒体并根据实时流媒体协议(RTSP)工作。为响应在媒体服务器上存贮消息的呼叫用户，媒体服务器还可提供给被叫用户可得到消息的通知。通知可采取电子邮件传送的形式，所述电子邮件包括被叫用户可检索该消息的地址。或者，通知可包括作为文件附件的消息拷贝。
根据本发明的另一实施例是一种提供统一通信服务的网络通话系统，所述网络通话系统包括至少两个分布在系统内的媒体服务器并适应单一通信服务器工作。
根据本发明的另一提供统一通信服务的网络通话系统包括数字数据网络和介于公共交换通话网络(PSTN)和数字据数据网络之间的网关服务器计算机。发送信号服务器可操作地连接到数字数据网络上。通信服务器也可操作地连接到数字数据网络上并响应发送信号服务器。还可提供包括用于存贮消息文件的计算机数据存贮媒体的媒体服务器。在网关服务器和媒体服务器之间有DTMF转换器，以便利用DTMF信号(如标准的电话手机)访问存贮在媒体服务上的消息文件。
根据本发明在数据网络通话系统中还提供了通信服务方法。所述方法包括接收从第一用户代理到第二用户代理的呼叫请求，其中用户代理为通话结点或端点。在响应呼叫请求中，总体上由发送信号服务器产生其它的呼叫请求，即到第二用户代理和到与第二用户代理相关的通信服务器的呼叫请求。根据探测到的消息条件，所述方法在第一用户代理和媒体服务器之间建立一个链接，以便提供在媒体服务器上的消息存贮。若没有探测到消息条件，取消到通信服务器上的呼叫请求。
消息条件可以在呼叫请求产生后预定时间内期满。消息条件也可以根据注册信息或与被呼叫方有关的状态信息(如请勿打扰指示)，或一个命令或按将特定呼叫转发到通信服务器的规则提供。
所述方法还提供呼叫恢复。在这方面，当呼叫用户正在媒体服务器上录制消息时，可向被呼叫用户提供邀请请求。若接受，被呼叫用户可进入呼叫通话并终止录制。
附图简述
为了全面理解本发明及其优点，下面结合附图进行描述，在所述附图中相同的标号表示相同的特点，其中：
图1为在SIP协议通话系统中提供统一通信服务的系统的框图；
图2为描述了根据图1所示系统提供的语音邮件服务例子的流程图；
图3为用于选择呼叫到通信服务器的路由的典型呼叫程序语言(CPL)脚本；
图4为用于适合与本发明系统和方法相连的消息管理系统的用户界面的图形表示；和
图5为根据本发明的系统的框图，包括互联网通话系统和常规PSTN之间的网关。
发明详述
图1为简化的框图，示出了实现有关互联网通话系统的统一通信服务的本系统的系统结构。所示的系统示出了部分局部通话交换或PBX的部分。所述系统优选根据用于发送信号和控制功能的通话启动协议(SIP)操作。所述系统总体上包括大量通话端点，所述通话端点优选采用SIP协议用户代理。出于图示目的，仅示出了两个这样的用户代理102、104。用户代理102、104可采用多种形式，如独立SIP通话装置，所述独立SIP通话装置可有多种获得方法或在常规个人计算机上运行的SIP客户端软件，如得到哥伦比亚大学，纽约市，纽约州许可的SIPC软件。在国际专利出版物WO00/76158于2000年12月14日公开的题目为“用于互联网/内联网通话的网络通话应用和系统”中描述了适当的SIP用户代理，通过引用其全部内容将其合并到本申请中。
SIP用户代理102、104连接到网络106上，如以太网。网络也可以是公共域下的具有用户代理群的互联网。用户代理102、104经由网络106(内部，对等网络)彼此直接连通，或从常规通话系统经由SIP-PSTN网关(外部)(图3)连通，或从外部另一互联网域连通。
所述系统总体上包括发送信号服务器108，其用来响应来自SIP用户代理102、104的呼叫请求并判别被叫用户的位置。发送信号服务器108优选为SIP服务器，其可执行代理和重发信号操作。在SIP协议中，每一通话端点可代表节点并具有特定的SIP地址。利用所述特定地址，作为呼叫用户的任何节点可直接与网络上的其它节点启动呼叫通话。发送信号服务器108可访问网络上的各种用户代理102、104，以便提供增强服务，如目录服务、呼叫转送、呼叫分发、呼叫通信及其类似的服务。例如，希望启动一个呼叫到JOHN SMITH的呼叫用户若知道个人SIP地址，则进入，如sip:john.smith@work.com。另一方面，若呼叫不知道被呼叫用户的SIP地址，呼叫用户可请求联系发送信号服务器108，请求与JOHN SMITH开始通话。
发送信号服务器包括有关不同用户注册信息的数据库，并能将SIP地址返回到呼叫用户或将呼叫请求转送至被叫用户的正确SIP地址。此外，被叫用户可有多个SIP地址，如john.smith@home，john.smith@office，john.smith@lab及其类似的地址。发送信号服务器可对每一个地址提供通话启动信号并在呼叫用户和第一个响应该启动请求的访问结点之间建立连接。同样，用户可用转发服务器定期注册，以表明可与他们联系(呼叫转移特点)的当前SIP地址。总的来说，当启动一个呼叫时，发送信号服务器108或代理、或转送或拒绝呼叫启动消息。发送信号服务器108，如已知的SIP代理服务器。适当的SIP代理服务器可利用由哥伦比亚大学，纽约市，纽约州许可的SIPD软件来实现相应服务。
系统100还包括经由网络106连接到发送信号服务器108上的通信服务器110。通信服务器110提供有关系统100的声音邮件和其它通信特性的命令和控制功能。当系统根据SIP协议运行时，通信服务器110包括SIP地址并根据协议运行。通常，通信服务器110对发送信号服务器108生成的对呼入或请求访问或管理消息作出响应的命令作出响应。
除了通信服务器110之外，还提供至少一个(示出了两个)单独的媒体服务器112A、112B。与提供控制功能的通信服务器110相比，尤其在涉及呼叫的路由中，媒体服务器112A、112B主要操作是：对呼叫者提供提示和呼出信息，记录信息，提供接收到的消息的通知及在各种媒体格式中进行消息检索。对大量用户的多媒体消息的存贮可能涉及大量存贮容量和带宽。为了分散系统负荷，多媒体服务器112A、112B可以遍及系统而分布。多个媒体服务器112可响应单独通信服务器110。媒体服务器112优选支持流媒体容量和协议，如RTSP协议。
如图1的连接所示，媒体服务器是用于多媒体邮件和消息的存贮地址。多媒体服务器112A和112B可配置在给定的系统内(一般由基于SIP系统的域所限定)，以便分散系统负荷并提供系统的伸缩性。
可有若干种方法在媒体服务器112上进行消息检索。例如，可使用公知的基于媒体播放器的RTSP直接播放存贮在流媒体服务器112A上的所录制的声音消息。例如可使用URI如rtsp://rtsp.mediaserver.ext/useragent_name/inbox/message_#.au来检索由流媒体服务器中用户收件箱中message_#所识别的信息，所述流媒体服务器可由项目mediaserver.ext来识别。此外，用户代理可经额e-mail将信息发送到当前位置上。信息检索的一种优选方法是通过网页使用常规网页浏览器软件，如Netscape
Figure G2009101808908D0000081
或Microsoft
Figure G2009101808908D0000082
对用户信箱进行访问和管理。下面结合图4对这些内容进行图示和描述。
结合图2的流程图所示的典型呼叫通话，将描述图1的系统100的操作。参考图2，SIP用户代理1 102试图呼叫SIP用户代理2 104(步骤202)。当SIP协议准备直接对等呼叫时，假定呼叫通过网络106经发送信号服务器108。发送信号服务器108产生“双路转送”，通过所述“双路转送”呼叫请求路由到通信服务器110和用户代理2104的当前注册地址(步骤204)。用户代理2中的呼叫请求提供铃声或其它呼入请求的通知。若在步骤206中，SIP用户代理在预定时间t内回答，代理服务器108接收到呼叫被接收的信号并取消向声音邮件服务器110发送的呼叫请求。发送信号服务器108接着向用户代理1 102发出响应，以便以本专业人员所知的方式启动呼叫通话(步骤207)。
若在步骤206中用户代理2104不能够在预定的时间t(如10秒)内提供呼叫被接收的信号，消息条件发生并且通信服务器110产生启动请求一指定的媒体服务器112A(步骤208)。消息条件还可以基于用户代理的选择或有效性而发生。例如，用户代理可将发送信号服务器108注册为对特定呼叫不可用或对每天的特定时间不可用。启动请求包括用于欢迎消息的重放请求和用来录制到来声音邮件消息的独立请求。在向媒体服务器112A发送的启动请求后，通信服务器110向代理服务器108提供信号，以便接收呼叫(步骤210)。发送信号服务器108取消对用户代理2 104的呼叫请求(步骤212)并向用户代理1提供响应将用户代理1指向媒体服务器112A(步骤214)。一旦发送到媒体服务器112A，用户代理1的操作者可直接通过媒体服务器提供录制的问候(步骤216)留下信息。
根据消息的完成，呼叫者、用户代理1 102终止呼叫通话，发送信号服务器108为媒体服务器112提供信号，以便停止录制。通信服务器110如通过E-mail提供消息已经收到的通知(步骤218)。通常，通知提供了一个链接，如统一源信息定位(URL)，引导SIP用户代理2的操作者至存贮消息的媒体服务器的位置。通过提供这样的一个链接，而不是信息本身，用户可访问远程消息，如经由Internet访问媒体服务器112。此外，通过将消息存贮在媒体服务器上并提供一个到信息的链接，系统带宽和负荷可被最小化。为了重新呼叫消息，用户进入链接到适当的网页浏览器并访问存贮在媒体服务器112中的消息(步骤220)。或者，信息可打包成e-mail附件并转发到用户。
完成图2所描述的“双路转送”通信的另一种方法是，重新发送到通信服务器110可直接程序进入用户代理接收呼叫。另一种方法是使用脚本，如图3所示的呼叫处理语言(CPL)脚本。这种方法为用户提供了对呼叫路由的绝大多数的控制。例如，所编写的脚本可根据特定的时间、用户代理的状态(如忙)、呼叫者地址及其类似的特征来选择发送呼叫。在CPL脚本发送方法具有最高的灵活性的同时，通信服务器110和发送信号服务器108需要支持CPL以提供该增强功能。
图4示出了管理和检索在媒体服务器上存贮的消息的用户界面。邮箱管理系统通常将消息接收到用户可访问的文件夹，如“收件箱”，“发送项目”等通常e-mail管理程序中所使用的，如Microsoft
Figure G2009101808908D0000101
参照图4，每一用户有一个消息收件箱，其中可存贮包括文本、图像、视频和声频消息的多媒体信息。消息管理系统通常显示所接收信息的日期和发送者，并可以显示消息的其它相关信息(如主题和消息大小)的信息。在主题中，可提供消息类型或消息类型的扩展类型。适当的统一消息管理软件包括哥伦比亚大学，纽约市，纽约州许可的SIPUM软件。
图5为框图，示出了图1系统至常规PSTN通话系统的扩展。在网络106以外运行的通话结点，即常规通话端点或SIP用户代理，经由SIP-PSTN网关504对发送信号服务器108进行访问。SIP-PSTN网关504包括网络106的界面及常规PSTN系统502的界面。来自PSTN的呼叫者可辩别和接收经SIP-PSTN网关504注册到发送信号服务器108上的SIP用户代理的呼叫。在基于通话网络的SIP和常规PSTN系统之间的SIP-PSTN网关504的使用为本专业人员知晓。
为了访问和控制媒体服务器112，也可提供DTMF-RTSP转换器406。SIP-PSTN网关508将DTMF声音信号转换为数字信息包数据，如使用RTP协议，并在网络106上传输。RTP协议数据包传送到DTMF-RTSP转换器506，所述DTMF-RTSP转换器506从常规通话装置中接收RTP编码的DTMF发送信号声音并提供相应的通过媒体服务器112解码的RTSP协议格式的响应命令。例如，在PSTN系统502内的常规电话手机的用户可访问存贮在媒体服务器112上的收件箱并利用电话手机的键区输入一些命令来完成有限的管理功能。例如，用户可按下键#1来接听新消息，键#4来应答消息，键#7删除消息及键#9来保存消息，等等。当然，各种命令可指定各种条目并且这组命令仅是一个例子。
本系统的另一特点是呼叫恢复。呼叫恢复的功能是指：被叫用户可在消息录制过程后选取一个呼叫，及提供选择使双方终止录制并启动一个普通呼叫通话。再参考图1，在本系统结构中，消息录制是由远方用户代理操作的，因此，发送信号方法必须干涉呼叫用户代理和媒体服务器112的录制过程。利用SIP协议可提供若干方法来实现这个目的。例如，当呼叫发送到媒体服务器112后并且到用户代理2的原呼叫取消后，媒体服务器可发送INVITE命令到用户代理2 104。若用户代理2 104的操作者接收到INVITE，用户代理2可与媒体服务器112和呼叫用户进行三路通话。媒体服务器112产生SIP BYE命令后脱离这次通话。
本系统和方法提供了一种用于递送统一通信服务的网络通话系统结构。与发送信号服务器分开的媒体服务器的使用增强了系统的灵活性和可伸缩性。这样，一普通发送信号服务器与若干媒体服务器共同工作使系统负荷和带宽优化。趋向于“媒体中枢”的SIP协议的使用提供了一个开放的系统结构方法。
本发明参照特定的优选实施例进行了描述。可以理解的是，本领域所属技术人员对这些实施例进行的变化和改造，没有脱离后面权利要求所述的本发明范围和宗旨。

Claims (8)

1.一种在数据网络通话系统中提供通信服务的方法，包括：
接收从第一用户代理到第二用户代理的呼叫请求；
对第二用户代理产生一呼叫请求并对通信服务器产生与第二用户代理相关的的呼叫请求；
在为与媒体服务器进行消息交换而提供一消息条件时，在第一用户代理和媒体服务器之间建立一个链接；
若没有消息条件发生，终止向通信服务器的呼叫请求。
2.根据权利要求1所述的提供通信服务的方法，其中，消息交换还包括从第一用户代理录制消息和将消息作为消息文件存贮在媒体服务器上。
3.根据权利要求1所述的提供通信服务的方法，其中，消息交换还包括提供从媒体服务器到第一用户代理的消息文件。
4.根据权利要求1所述的提供通信服务的方法，还包括为消息有效的被叫用户提供通知。
5.根据权利要求1所述的提供通信服务的方法，还包括当第一用户代理录制消息时，邀请被叫用户恢复呼叫。
6.根据权利要求1所述的提供通信服务的方法，其中，消息条件是呼叫请求的一段预定时间。
7.根据权利要求1所述的提供通信服务的方法，其中，消息条件是与被叫用户相关的注册记录的入口。
8.根据权利要求1所述的提供通信服务的方法，其中，消息条件是被叫用户指定被叫用户无效的响应。
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