CN1421008A - 基于会话启动协议的特征控制 - Google Patents

基于会话启动协议的特征控制 Download PDF

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CN1421008A
CN1421008A CN00818232A CN00818232A CN1421008A CN 1421008 A CN1421008 A CN 1421008A CN 00818232 A CN00818232 A CN 00818232A CN 00818232 A CN00818232 A CN 00818232A CN 1421008 A CN1421008 A CN 1421008A
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eigenstate
information
message
sip
feature
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S·R·多诺万
R·德瓦纳坦
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Verizon Business Global LLC
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MCI Worldcom Inc
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    • 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
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • H04L65/1104—Session initiation protocol [SIP]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/42314—Systems providing special services or facilities to subscribers in private branch exchanges
    • 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
    • H04L65/1066—Session management
    • H04L65/1096—Supplementary features, e.g. call forwarding or call holding
    • 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
    • H04L65/1066—Session management
    • H04L65/1101—Session protocols
    • 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
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L69/00—Network arrangements, protocols or services independent of the application payload and not provided for in the other groups of this subclass
    • H04L69/30—Definitions, standards or architectural aspects of layered protocol stacks
    • H04L69/32—Architecture of open systems interconnection [OSI] 7-layer type protocol stacks, e.g. the interfaces between the data link level and the physical level
    • H04L69/322—Intralayer communication protocols among peer entities or protocol data unit [PDU] definitions
    • H04L69/327—Intralayer communication protocols among peer entities or protocol data unit [PDU] definitions in the session layer [OSI layer 5]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M3/00—Automatic or semi-automatic exchanges
    • H04M3/42—Systems providing special services or facilities to subscribers
    • H04M3/436—Arrangements for screening incoming calls, i.e. evaluating the characteristics of a call before deciding whether to answer it

Abstract

公开了一种用于提供对Session Initiation Protocol(会话启动协议)的一个补充的系统和方法。该补充是一个新的字段标题，最好是取名为“Feature(特征)”，它被增加到REGISTER(1)消息上。这个字段将包含用于各种特征服务、如同由传统的PBX系统提供的Do Not Disturb特征及其它服务的控制信息。

Description

基于会话启动协议的特征控制
发明背景技术
1.发明领域
本发明通常涉及国际互连网多媒体和单媒体通信，尤其是涉及一种用于使用Session Initiation Protocol(会话启动协议)提供基于特征的服务的方法和系统。
2.相关技术描述
已经看到最近二十年来国际互连网按指数规律地增长。从一个用于技术了解的工具来看，国际互连网已经变成一大部分普通大众的一个主要通信方式。随着它的不断增长，其它工具和技术、诸如音乐和视频已经移植到基于计算机的介质上。
移植技术的一个实例是电话服务向国际互连网的移动。国际互连网电话通信是指各种各样的服务，不仅仅是传统的电话呼叫。通常，它是指实时媒体、诸如语音和视频经由国际互连网的传输以在国际互连网用户之间提供交互式通信。用户可以通过各种装置-个人计算机(PC)、独立的网际协议(Internet Protocol，IP)启动的设备、甚至通过从一个传统的公共交换电话网(Public Switched TelephoneNetwork，PSTN)的电话拨号到一个网关来访问国际互连网。
IP电话通信的优点是相当广泛的。它提供了高质量的语音通信、改善了的多路复用增益、丰富的计算机电话通信集成、高级的服务、用于供应者的一个开放性市场、以及降低了的成本。大部分这些优点来自于下列事实：IP电话通信是基于数据包的，而不是像现在的电话系统那样基于电路的。在一个基于电路的通信、诸如在传统设备上的一个电话呼叫中，在呼叫者和被呼叫者之间通过交换机和导线保持一个实际连接。在一个基于数据包的通信中，该通信本身被分解为经由一个网络发送的数据包。这允许资源的一个更有效利用。此外，IP电话通信大多使用基于文本的协议，它允许诸如Java、Tcl、和Perl之类语言的容易实现和调试。在IP电话通信中的分层协议也被设计为彼此相互独立操作。这意味着：一个人能够在很宽的设备范围内、从蜂窝和台式电话到TVs、立体声、和计算机中，使用相同的较高层协议，这是因为较低的物理层将负责处理这些差异。
当前，传统的PSTN电话通信使用SS7(System Signaling 7)作为信令协议，它建立、控制、和断开电路连接。IP电话通信使用SIP(Session Initiation Protocol，会话启动协议，RFC 2543)作为用于电话通信、视频会议、和多媒体会话的信令协议。SIP提供了建立和释放连接。
SIP协议是一个基于文本的协议，它在TCP/IP(Transport ControlProtocol/Internet Protocol，传输控制协议/网际协议)栈中的传输层上工作。SIP能够使用任何传输协议、包含TCP(TransportControl Protocol，传输控制协议)和UDP(User Datagram Protocol，用户数据报协议)作为它的传输协议。此外，SIP还可以利用ATM AAL5(Asynchronous Transfer Mode ATM Adapt ion Layer 5，异步传输模式ATM适应层5)、IPX(Internet Packet eXchange，网际包交换)、帧中继或X.25传输协议工作。
在一个SIP网络中有两个组分：网络服务器和用户代理。用户代理是一个以想参与呼叫的某人的名义进行动作的终端系统。通常，用户代理包含：一个启动一个呼叫的协议客户(一个用户代理客户UAC)，和一个对一个呼叫作出响应的协议服务器(用户代理服务器UAS)(参见图1)。同样有两种不同类型的网络服务器：一个代理服务器，它接收请求，确定把它发送到哪个服务器，然后转发该请求；以及一个重定向服务器，它接收请求，但不是把它们转发到下一个中继服务器，而是告诉该客户直接与该下一个中继联系。
在启动一个会话过程中的步骤是相当简单的：如图1所示，(1)UAC发送一个INVITE请求到一个SIP服务器，在这种情况下该SIP服务器是一个代理服务器。该SIP服务器将搜索它的数据库以确定把该INVITE请求发送到哪里。一旦确定了，该代理服务器就把INVITE消息发送到适当的下一个中继。在图1中，下一个中继就是被呼叫者，但是，实际上在SIP服务器和被呼叫者之间能有许多中继。如果SIP服务器是一次重定向服务器，则它将向UAC通知适当的下一个中继是什么，并且使UAC执行其余部分。一旦(2)INVITE消息最终到达被呼叫者UAS，(3)被呼叫者UAS用一条OK消息做出响应，然后(4)该OK消息被转发到呼叫者UAC。当呼叫者UAC接收了该OK消息时，表明被呼叫者已经接收了该INVITE，(5)UAC发送一条ACK消息，(6)当该ACK消息被接收时将开始会话。
然而，在把当前所有传统电话网络中的服务移植到国际互连网电话通信和SIP中存在着困难。在传统的网络中，Private Branch Exchange(用户交换机，PBX)是商业语音通信连网的基石。例如，一个公司中的所有雇员将把在他们办公桌处的电话连接到一个PBX，由它路由选择呼叫、处理语音邮件、并且提供其它服务。用基于IP(InternetProtocol，网际协议)的PBXs替换传统的PBXs将要求：用户从他们IP启动的办公桌电话中得到的服务与他们以前从他们的传统PBX电话中得到的服务相同。
传统PBXs提供的服务类型的一个实例是Do Not Disturb(请勿打扰)特征。Do Not Disturb(DND)经常被实现作为一个按钮，用户按下它之后使得所有呼叫都将被自动地转发到用户的语音信箱。在一个传统的PBX中，一条DND命令信息通常从台式设备发送到该PBX，然后由PBX自动地转发所有呼叫到语音邮件系统。但是，在一个基于SIP的IP电话网络中，办公桌电话、或更确切地说是用户代理服务器UAS将执行转发。尽管这在很多情况下进行，但是它假定终端客户设备(在这种情况下，基于SIP的IP办公桌电话)具有一个DND性能，而这不是确定的。因此，需要在终端客户设备、诸如基于S1P的IP办公桌电话中有一种用于传统PBX服务、诸如DND特征的系统和方法，而不必假定该终端客户设备具有这个性能。
此外，在传统PBX系统中有其它特征，它们要求向网络服务器通知状态变化，特别是当那些变化影响了一个用户已经签署了的、一个基于网络的服务的会话/呼叫启动处理时。例子包含：当用户希望把所有的呼叫转发到另一个号码/地址时，无条件的呼叫转发；当用户希望在一定条件下转发某些呼叫时，有条件的呼叫转送；当用户希望阻塞一定的地址/号码被呼叫时，呼叫阻塞；以及，当用户希望阻止来自于一定地址/号码的呼叫被接收时，呼叫屏蔽。因此，需要有一种用于使用户代理通知网络服务器特征状态改变、诸如DND、呼叫转发、屏蔽以及阻塞等的方法和系统。
发明概述
本发明的一个目的是提供一种允许在一个基于SIP的电话通信网络中的网络服务器执行一个传统PBX系统的所有功能的系统和方法。
本发明的另一个目的是允许传统的PBX功能在一个网络中执行，而不必要求在那个网络中的终端客户设备能够执行所有这些功能。
本发明另外一个目的是提供一种用于向网络服务器通知一个特定用户或设备的特征信息改变的系统和方法。
为了实现这些及其它目的，本发明提供了对SIP协议的一个扩展，它将被用于向网络服务器通知用于一个特定用户或设备的特征信息已经改变了。由一个额外的被称作“Feature”的字段标题组成的这个扩展将允许一个终端客户机向网络服务器通知特征改变，而不是执行这些特征本身。
附图简要说明
通过下列详细描述并且结合附图，本发明的上述及其它目的、特征、方面和优点将变得更为明显，其中：
图1显示了一个基于SIP的系统的组件以及启动一个会话的概述；以及
图2显示了依据本发明一个实施例的REGISTER消息的使用。
最佳实施例的详细说明
在一个传统的PBX系统中，当终端用户想要修改或开始使用一个特征、诸如DND时，电话单元将用指示该特征控制信息的专用数字发送一条消息。例如，DND按钮可以发送一条“*720”消息，它通知PBX在那个电话扩展上激活DND特征。释放DND按钮可以发送一条“@720”消息，它通知PBX在那个电话扩展上使DND特征无效。
本发明的最优实施例使用已经存在于SIP中的一个消息类型(REGISTER)以发送这类特征控制信息。为了这样做，该最佳实施例添加另一个字段标题、“Feature”字段标题到REGISTER消息上。
为了更充分地理解该最佳实施例，接下来简要论述在SIP中的各种类型的方法消息。INVITE和ACK是以上所提及的两种方法消息；其它方法消息包含OPTIONS、BYE、CANCEL、和REGISTER。INVITE消息指示该用户或服务正在被邀请参与一个会话。ACK消息证实该客户机已经接收了对一条INVITE消息的一个最终响应。OPTIONS消息是对服务器的一个涉及服务器性能的查询。BYE消息由客户机使用以向该服务器指示它希望释放或结束该呼叫。CANCEL消息被用来取消一个待处理请求。
REGISTER消息能够被理解为一条“登录”消息，尽管它具有更多的使用。举例来说，办公桌电话能够被设计为具有一个“I am here(我在此)”按钮，用于向SIP服务器指示实际上那个人在那个电话处并且准备好接收电话呼叫及其它多媒体通信。这个“I am here”按钮将发送一名REGISTER消息到SIP服务器。它可以或可以不用一条请求从SIP服务器批准的请求进行追踪以确保保密性。如果它全部解决了，则SIP服务器将在SIP服务器数据库中记录那个用户和该用户的地址，从而使呼叫能够被适当地路由。
通常，登录为由SIP服务器提供的用户服务批准一个SIP用户代理或者使其无效。另外，用户代理提供一个或多个到SIP服务器的联系位置。一条REGISTER消息的一个例子如下，一个名为“Barney”的用户正使用它以登录到一个SIP服务器：
REGISTER sip：ss2.wcom SIP 2.0
Via：SIP/2.0/UDP there.com：5060
From：Barney<sip：UserB@there.com>
To：Barney<sip：UserB@there.com>
Call-ID：123456789@there.com
CSeq：1 REGISTER
Contact：Barney<sip：UserB@there.com>
Contact：sip：+1-972-555-2222@gwl.wcom.com；user＝phone
Contact：tel：+1-972-555-2222
Authorization：Digest username＝″UserB″，realm＝″MCIWorldCom SIP″，
Nonce＝″ea9c8e88df84flcec4341ae6cbe5a 359″，opaque＝″″，
uri＝″sip：ss2.wcom.com″，
response＝″dfe56131d1958046689cd83306477ecc″
Content-Length：0
标题字段的一个确切描述被包含在RFC 2543中，但是这种细节对完全理解本发明来说是不必要的。然而，作为一个旁白，Authorization字段标题被用来确认一个授权用户正进行登录。
在本发明的一个最佳实施例中，另一个字段标题被增加到REGISTER消息上，它将传送特征控制信息。这将允许REGISTER消息执行模拟一个传统PBX系统所要求的附加功能。例如，如果在一个基于IP的PBX系统中的用户按下了DND按钮，则从该终端客户机设备、即电话发送一条适当格式化了的REGISTER消息到SIP Proxy Server(SIP代理服务器)，如图2中的(1)所示。REGISTER消息将具有下列格式(添加的粗体表示新的标题)：
REGISTER.sip：ss2.wcom SIP 2.0
Via：SIP/2.0/UDP there.com：5060
From：Barney<sip：UserB@there.com>
To：Barney<sip：UserB@there.com>
Call-ID：123456789@there.com
CSeq：1 REGISTER
Contact：Barney<sip：UserB@there.com>
Contact：sip：+1-972-555-2222@gwl.wcom.com；user＝phone
Contact：tel：+1-972-555-2222
Authorization：Digest usemame＝″UserB″，realm′＝″MCIWorldCom SIP″，
nonce＝*ea9c8e88df84flcec4341ae6cbe5a359″，opaque-″″，
uri＝″sip：ss2.wcom.com″，
response＝dfe56131d1958046689cdB3306477ecc″
Feature＝dnd；status＝active
Content-Length：0
这个Feature字段入口将通知SIP服务器改变该“dnd”特征状态为“有效的”。因此，SIP服务器将知道发送所有Barney的进来呼叫到Barney的语音邮箱。这允许用户代理客户机UAC是“薄的”；换句话说，用户代理客户机将不必增加所有的PBX特征到它自己的协议中。相反，用户代理客户机(在我们的例子中，是一个基于IP的办公桌电话)将简单地通知网络服务器以提供该特征。当SIP服务器已经适当地更新了它的表格和数据库时，SIP服务器发送(2)一条OK消息到用户代理客户机UAC以证实该特征状态信息已经被更新了。OK消息将具有下列格式(添加的粗体表示新的标题)：
        SIP 2.0 200 OK
Via：SIP/2.0/UDP there.com：5060
From：Barney<sip：UserB@there.com>
To：Barney<sip：UserB@there.com>
Call-ID：123456789@there.com
CSeq：1 REGISTER
Contact：Barney<sip：userB@there.com>
Contact：sip：+1-972-555-2222@gwl.wcom.com；user＝phone
Contact：tel：+1-972-555-2222
Feature＝dnd；status＝active
Content-Length：0
当(3)正在使用一个SIP启动的客户机设备的某人、诸如Frank试图呼叫Barney时，SIP服务器将参考它被更新了的表格和数据库，并且发现Barney希望他的消息被输入到语音邮件系统中。SIP服务器于是将(4)重定向进来呼叫到语音邮件消息传送系统。然后(5)语音邮件系统将发送一条OK消息到Frank以便启动一个连接。
当使用诸如呼叫转发、屏蔽或阻塞之类的特征时，Feature字段入口将包含附加信息。例如，如果被激活了的特征是呼叫屏蔽，则Feature字段入口能够表现为：
Feature＝screen；status＝active；tel：+1-972-555-1212
包含这个Feature字段入口的REGISTER消息将指示SIP服务器激活呼叫屏蔽并且防止电话号码1-972-555-1212呼叫该用户。明显地，其它特征可能要求或多或少的信息，这取决于SIP服务器将被要求执行的功能。
由该最佳实施例提供的解决方案与诸如在一个PBX系统中执行的传统消息处理相比较，其更容易实现，而且更容易理解。此外，“dnd；status＝active”意味着什么与传统的“*720”相比较在直观上更加明显。
虽然本发明已经结合最佳实施例进行了详细描述，但是它仅仅是一个示范性的应用。因此，显然应当明白，本领域的技术人员能够在本发明的范围和精神内做出许多变化。

Claims (20)

1.一种用于在一个基于会话启动协议(SIP)的网络中提供特征状态信息的系统，所述网络具有包含至少一个字段标题的消息，该系统包含：
用于把特征状态信息作为一个附加字段标题增加到一条消息上的装置。
2.如权利要求1所述的系统，进一步包含：
由一个用户操作以启动一个会话的一个客户机装置，所述客户机装置发送和接收来自于一个服务器装置的请求；以及
一个服务器装置，用于接收一个请求，确定把一个请求发送到哪里，以及转发一个请求；
其特征在于：包含所述特征状态信息的所述消息在客户机装置和服务器装置之间进行发送。
3.如权利要求1所述的系统，进一步包含：
由一个用户操作以启动一个会话的一个客户机装置，所述客户机装置发送和接收来自于一个服务器装置的请求；以及
一个服务器装置，用于接收一个请求，确定把该请求发送到哪里，并且通知所述客户机装置把该请求发送到那里；
其特征在于：包含所述特征状态信息的所述消息在客户机装置和服务器装置之间进行发送。
4.如权利要求1所述的系统，进一步包含：
一个SIP服务器，用于提供特征并且接收具有特征状态信息的消息；
其特征在于：所述SIP服务器依据包含在所接收消息中的特征状态信息实现该特征。
5.如权利要求1所述的系统，其特征在于：包含所述特征状态信息的所述附加的字段标题被添加到一个REGISTER消息上。
6.如权利要求1所述的系统，其特征在于：所述基于SIP的网络启动电话通信会话。
7.如权利要求6所述的系统，其特征在于：对应于该特征状态信息的一个特征正在呼叫转发。
8.如权利要求6所述的系统，其特征在于：对应于该特征状态信息的一个特征正在呼叫阻塞。
9.如权利要求6所述的系统，其特征在于：对应于该特征状态信息的一个特征正在呼叫屏蔽。
10.一种用于在一个基于会话启动协议(SIP)的网络中提供特征状态信息的方法，所述网络具有至少一个用户代理客户机(UAC)、用于实现特征的至少一个SIP服务器，以及包含至少一个字段标题的消息，该方法包含以下步骤：
增加特征状态信息到一条来自UAC到SIP服务器的消息上；
传输所述消息到SIP服务器；以及
依据在所述消息中的特征状态信息实现所述特征。
11.如权利要求10所述的方法，其特征在于：所述UAC是在由一个用户使用的一个终端客户机设备中实现的，而且该方法进一步包含以下步骤：
向该终端客户机设备指示用户希望进行什么特征状态改变；以及
由所述指示生成特征状态信息。
12.如权利要求10所述的方法，其特征在于：所述特征状态信息具有一个特征状态字段标题。
13.如权利要求10所述的方法，其特征在于：具有增加的特征状态信息的所述所述消息是一条REGISTER消息。
14.如权利要求10所述的方法，其特征在于：所述基于SIP的网络启动电话通信会话。
15.如权利要求10所述的方法，其特征在于：对应于该特征状态信息的一个特征正在呼叫转发。
16.如权利要求10所述的方法，其特征在于：对应于该特征状态信息的一个特征正在呼叫阻塞。
17.如权利要求10所述的方法，其特征在于：对应于该特征状态信息的一个特征正在呼叫屏蔽。
18.一种用于在一个基于会话启动协议(SIP)的网络中提供特征状态信息的系统，所述网络具有包含至少INVITE、ACK、OPTIONS、BYE、CANCEL、REGISTER的方法消息，其中每一条所述方法消息具有至少一个字段标题，该系统包含：
至少一个用户代理客户机，用于使用一个附加的字段标题以增加特征状态信息到一条REGISTER消息，所述用户代理客户机发送所述REGISTER消息到至少一个SIP服务器；
所述至少一个SIP服务器，用于从所述至少一个用户代理客户机接收所述REGISTER消息，并且依据在所述REGISTER消息中的特征状态信息实现特征；以及
至少一个所述网络，用于在所述至少一个用户代理客户机和所述至少一个SIP服务器之间传输消息。
19.如权利要求18所述的系统，其特征在于：所述网络启动电话通信会话。
20.如权利要求19所述的系统，其特征在于：所述特征包含呼叫转发、呼叫屏蔽、和呼叫阻塞中的一个。
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