CN1279734C - 用于VoIP呼叫控制的SIP消息的区别处理 - Google Patents
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/215—Flow control; Congestion control using token-bucket
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/24—Traffic characterised by specific attributes, e.g. priority or QoS
- H04L47/2408—Traffic characterised by specific attributes, e.g. priority or QoS for supporting different services, e.g. a differentiated services [DiffServ] type of service
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/24—Traffic characterised by specific attributes, e.g. priority or QoS
- H04L47/2416—Real-time traffic
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/24—Traffic characterised by specific attributes, e.g. priority or QoS
- H04L47/2441—Traffic characterised by specific attributes, e.g. priority or QoS relying on flow classification, e.g. using integrated services [IntServ]
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L47/00—Traffic control in data switching networks
- H04L47/10—Flow control; Congestion control
- H04L47/24—Traffic characterised by specific attributes, e.g. priority or QoS
- H04L47/2475—Traffic characterised by specific attributes, e.g. priority or QoS for supporting traffic characterised by the type of applications
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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
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- 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]
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Abstract
本发明提供一种根据VoIP呼叫控制消息的重要性和功能性,区别处理VoIP呼叫控制消息,从而向VoIP基础结构提供和PSTN网络类似的稳健呼叫控制水平的方法。本发明依据SIP消息的类型或内容,对SIP消息分类,并通过按照策略,向它们分配VoIP呼叫控制服务器的资源。该方案还提供VoIP呼叫控制服务器中某些类型请求的过载保护和优先处理。
Description
技术领域
本发明涉及计算机网络领域。更具体地说,涉及使用会话起始协议(SIP)的基于因特网协议的话音(VoIP)网络。
背景技术
传统的电话载体(公共交换电话网或PSTN)正在朝着基于分组的VoIP基础结构发展。任何电话基础结构的关键部分是呼叫控制。呼叫控制包括呼叫建立和呼叫断开(teardown)。在VoIP中,可通过SIP实现呼叫控制。SIP是VoIP消息的标准协议。SIP消息是基于文本的消息。
稳健的VoIP呼叫控制基础结构的关键要求是过载控制。由于VoIP是新兴领域,基础结构的不同组件目前处于试验阶段,还未进行VoIP的大规模部署。一旦开始大规模部署,VoIP基础结构中的呼叫控制将变成避免服务中断和收益损失方面的关键。由于VoIP是新兴领域,关于VoIP服务器的过载保护的工作很少。
除了陆线电话服务之外,SIP还被选为第三代(3G)移动电话服务的信令机制,具体地说由3GPP标准团体在其release UMTS R00中被选为第三代(3G)移动电话服务的信令机制。这种发展的含义是陆线和移动电话服务都需要基于SIP的呼叫控制基础结构的过载保护。第二种含义是在移动电话服务中,控制基础结构必须不仅支持呼叫建立和断开,而且控制基础结构还必须处理不同小区边界之间的终端移动性。对越区切换的这种支持代表对呼叫控制服务器的额外负载。
另一种重要的发展是使用SIP传送即时消息(IM)和支持“在席”(Presence)。“在席”是本领域的普通技术人员明白的一个术语,指的是使***中的感兴趣的用户知道谁在线/离线的概念。例如,在某人登录之后,SameTime服务器显示“绿”灯。两种方法目前正在IETF标准团体进行标准化。称为SIMPLE的一种方法在SIP消息中传送即时消息接发有效负载,当网络朝着提供集中的语音和数据服务的方向发展时,可以想象的是,SIP消息将由相同的呼叫控制基础结构处理。把SIP用于即时消息接发和在席意味着对基于SIP呼叫控制基础结构的额外加载。
除了处理呼叫建立、断开、越区切换和即时消息接发之外,SIP呼叫控制服务器还需要与其它呼叫控制服务器交换路由信息,以便恰当地路由呼叫建立消息。
出于上述原因,本领域需要一种关于VoIP的SIP消息的呼叫控制方法和***。
发明内容
于是,本发明的一个目的是在呼叫控制服务器处理消息之前,对SIP消息分类,并把它们放入不同队列中,从而能够根据消息的重要性和功能性实现服务器资源的分配,而不是按照先进先出的到达顺序处理消息。
本发明的第二目的是以SIP消息中各个字段的值为基础的SIP消息的基于文本的分类。
本发明的第三目的是根据消息的预定功能,对SIP消息分类。所述功能可包括终端注册(SIP REGISTER),呼叫建立(SIP INVITE),呼叫断开(SIP BYE),终端移动性(SIP RE-INVITE),和作为SIP消息中的有效负载传送的即时消息接发。
本发明的第四目的是根据SIP消息的预定功能,对保持SIP消息的各个队列应用不同的服务速率。
本发明的第五目的是在基于分组的语音基础结构中,向紧急呼叫(911)提供最高优先权。
结合附图,根据本发明的下述详细说明,本发明的这些及其它目的、特征和优点将变得显而易见。
附图说明
图1图解说明现有技术的SIP消息的处理;
图2图解说明根据本发明一个实施例的SIP消息的处理。
具体实施方式
图1图解说明根据现有技术的SIP消息队列100的处理。控制器201简单地响应每个SIP控制消息101-104。例如,SIP消息可以是蜂窝电话***中的呼叫越区切换请求。控制器201接收SIP消息,并据此命令与该消息相关的呼叫的越区切换。
在SIP消息104与911呼叫相关的情况下,现有技术只提供SIP控制消息的先进先出处理。如果处理SIP消息101-103的延迟足够大,那么由于对越区切换请求的响应缓慢,911呼叫可能丢失。
本发明允许SIP呼叫控制消息的区别处理。本领域的普通技术人员会认识到关键的SIP消息可包括INVITE(会话起始),reINVITE(会话修改),REGISTER(把某一用户和某一地址绑定)和BYE(会话结束)。reINVITE和REGISTER的组合可被用于处理用户/终端移动性。可按照策略规范,对到来的请求分类和标记。这些请求可被认为类似于web请求,并且可与资源的隔离一道,按照优先化策略处理,以便处理各种消息。
图2图解说明了本发明的***和方法的一个实施例。本发明可被看作接收SIP呼叫控制消息流200的软交换机(soft switch)400。流200也可被称为SIP服务器队列。软交换机是本领域的普通技术人员明白的术语,它可包括SIP代理,呼叫控制,用户注册等。软交换机实质上包含传统上通过电话交换机完成的角色。交换机400可包括一个分类器420;许多权标桶(token bucket)421-423;一个控制器401;和一个CPU调度器。
分类器420可体现为根据消息的类型,分离到来的消息的软件模块,微控制器,或者过滤器。由于SIP消息是基于文本的消息,因此不仅可以根据SIP对到来的消息分类,而且还可根据应用的具体类型,对到来的消息分类。分类器420也可靠未来定义的关于通信建立的其它信令协议来工作。
权标桶421-423可以实现成按照重要性或优先化,等待由控制器401处理的队列。控制器401从到来的消息读取呼叫控制请求,并据此执行呼叫交换功能。
在图解说明的实施例中,已知队列的优先化,CPU调度器403实现控制器401调度策略。控制器401表示在与交换机相关的CPU上运行的线程,CPU调度器403根据调度策略,调度线程以便执行。例如,调度器403可把50%的处理时间分配给队列421,把30%的处理时间分配给队列42,把20%的处理时间分配给队列423。控制器401随后据此处理请求。仍然参见图2,下面说明本发明的方法的一个实施例。
本发明的体系结构包括对SIP服务器队列的入站通信量控制,以便调节SIP消息的处理,如上所述。这些消息经过警管器(policer)(例如分类器420),所述警管器通过对到来的消息流强制执行漏桶流量控制,能够实现消息过载保护。标记的消息进入***400,并竞争处理资源。通过恰当地调度各种类别的工作负荷,向它们提供服务区别和QoS保证。对于调度器来说,各种消息上的标记是可见的,允许它们按照每种工作的性能要求,不同地处理每种工作。
如上所述,由于SIP消息是基于文本的消息,因此不仅能够根据SIP消息类型对到来的消息分类,而且能够根据具体的应用类型,对到来的消息分类。例如,911呼叫(会话)可被给予最高优先权(例如放入具有50%的处理时间的队列421中)。类似地,需要通过独立的域路由呼叫的SIP INVITE消息(例如MCI呼叫者到AT&T被叫者)可服从第二域已预协商的关于呼叫速率的双边对等安排。从而,和要求SIP消息被转发给另一域的那些呼叫相比,可能需要对在相同域内终止的呼叫强制推行不同的策略(速率)。通过与正常的固定线路会话(呼叫)相对,处理移动呼叫,推动资源划分(计算/呼叫处理)的另一要求。这应由处理SIP INVITE消息和SIP reINVITE消息的队列的权标桶速率确定。使用域间软交换信令(SIP BCP-T)的SIP消息是需要对其进行速度控制和确保呼叫处理资源的最小共享的另一组消息。
下面表示了一个例证的SIP INVITE消息。该例子突出了SIP协议的文本本质,SIP协议的文本本质为基于消息内容以及消息类型的SIP消息的服务区别创造条件。
INVITE:sip:
[email protected] SIP/2.0
Via:SIP/2.0/UDP 12.26.17.91:5060
Max-Forwards:70
To:Bob<sip:[email protected]>
From:Alice<sip:[email protected];tag=1928301774>
Call-ID:[email protected]
CSeq:314159 INVITE
Contact:<sip:[email protected]>
Content-Type:application/sdp
Content-Length:142
本例中,该消息代表一个新的呼叫建立请求,因为它是一个INVITE消息。分类器420将把它放入处理新的呼叫建立的队列中。
从而,本发明包括一种区别服务的交换方法,该方法向VoIP网络提供迄今为止,只有在公共交换电话网(PSTN)中才能获得的服务水平保证。
虽然关于一个实施例,说明了本发明,不过本领域的技术人员会认识到在附加权利要求的精神和范围内,可对本发明做出各种修改。另外,申请意图包含所有权利要求(即使稍后在申请的处理过程中被修改)的等同物。
Claims (6)
1、一种方法,用于处理基于因特网分组的语音的呼叫控制的会话起始协议SIP消息,包括:
接收SIP消息流;
根据至少两种消息类型,对消息分类;
把所述消息放入与消息类型相关的不同队列中;和
根据预定策略,把SIP呼叫控制服务器处理资源分配给每个队列。
2、按照权利要求1所述的方法,其中对消息分类的步骤包括把消息分类为REGISTER、INVITE或reINVITE消息。
3、按照权利要求2所述的方法,其中对消息分类的步骤包括通过读取SIP INVITE消息的目的地地址,把消息分类为紧急呼叫消息。
4、按照权利要求1所述的方法,其中分配资源的步骤包括通过使用用于处理各个队列的权标桶速率控制把不同程度的服务器处理资源分配给SIP消息的各个队列;每个队列的权标生成的所述速率由附属于消息类型的重要性指示。
5、按照权利要求1所述的方法,其中分配资源的步骤包括控制呼叫控制服务器处理来自各用户的消息的速率,从而在基于分组的基于因特网协议的语音VoIP基础结构中防止各服务器对呼叫控制服务器的拒绝服务攻击。
6、一种***,用于处理基于因特网分组的语音的呼叫控制的会话起始协议SIP消息,包括:
接收SIP消息流并根据至少两种消息类型对消息分类的分类器;
与消息类型相关的多个队列;
在队列中定向与消息对应并等待被服务的呼叫的SIP控制服务器;和
根据预定策略,把SIP呼叫控制服务器处理资源分配给每个队列的调度器。
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US10/714,732 US7701854B2 (en) | 2003-11-17 | 2003-11-17 | Differentiated handling of SIP messages for VoIP call control |
US10/714,732 | 2003-11-17 |
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US7701854B2 (en) | 2010-04-20 |
CN1620046A (zh) | 2005-05-25 |
US20050105464A1 (en) | 2005-05-19 |
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