英语翻译The Session Initiation Protocol (SIP) is an application protocol,defined in RFC2543 [4],that is being designed by theIETF MMUSIC (Multiparty Multimedia Session Control) working group to enable users to participate in multimediasessions,th

英语翻译The Session Initiation Protocol (SIP) is an application protocol,defined in RFC2543 [4],that is being designed by theIETF MMUSIC (Multiparty Multimedia Session Control) working group to enable users to participate in multimediasessions,th
英语翻译
The Session Initiation Protocol (SIP) is an application protocol,defined in RFC2543 [4],that is being designed by the
IETF MMUSIC (Multiparty Multimedia Session Control) working group to enable users to participate in multimedia
sessions,that is,to establish,modify and terminate multimedia sessions calls.MMUSIC working group [5] focus on
loosely coupled conferences as they exists today on the MBONE.One of the main issues in this area is related with
how to inform users about forthcoming sessions,media requirements,addresses,etc.There are two basic ways to
locate and to participate in a multimedia session:
· Advertisement.Sessions are advertised in various ways like email,web pages,newsgroups or a multicast
advertisements via Sessions Announcement Protocol (SAP) like in the MBONE.
· Invitation.Users are invited by others to participate by using the Session Initiation Protocol (SIP).
SIP has been proposed as a generic unicast and multicast initiation protocol and also as an IP Telephony call set-up
protocol.It is based on a client-server protocol.SIP clients send a Request Message for a service,and a server
handles the request,answering with a Response Message.SIP terminals can both generate and receive request as
they are composed by a User Agent Client (UAC) and a User Agent Server (UAS).
SIP terminals can establish voice calls directly without requiring any other element.Figure 6 shows an example in
which user1 calls user2 by sending an INVITE Request primitive containing user1 supported capabilities for receiving
audio and a UDP port (port 12345 and mlaw codec).When user2 receives INIVITE Request,he can establish a voice
channel to 12345 UDP port of user1 while he accepts the request by sending an OK Response message.In addition,
user2 response includes its own media capabilities,which are used by user1 to establish a voice channel (GSM codec
at 54321 port in the example) and send an ACK message to acknowledge user2’s response.In order to terminate the
connection,any of the parties can send a BYE Message,which must be acknowledged by an OK response.

英语翻译The Session Initiation Protocol (SIP) is an application protocol,defined in RFC2543 [4],that is being designed by theIETF MMUSIC (Multiparty Multimedia Session Control) working group to enable users to participate in multimediasessions,th
会议开始协议(SIP)是一种应用协议,确定在rfc2543[4] 正在设计由IETF的MUSIC(多方多媒体会议控制)工作组,以帮助用户参与多媒体会议,就是要建立,修改和终止多媒体会议要求. mmusic工作组[5]集中松散耦合会议,因为他们今天存在对mbone. 的主要问题之一,在这方面是与如何向用户介绍即将举行的会议 媒体要求,地址等,有两个基本方法,寻找并参与了多媒体会议: ·广告. 会议的广告以各种方式如电子邮件,网页 新闻或广告组播会议通过公布协议(SAP)的一类的mbone. ·邀请. 用户邀请其他人参加用会话初始协议(SIP). 学校已提出作为一个通用组播入会议定书,并作为一个IP电话呼叫建立起来 议定书. 它是基于客户-服务器协议. SIP的客户发出请求信息服务,而服务器处理的要求,回答了回应讯息. SIP的终端既可以赚取和接受的要求,因为它是由一个用户代理客户(87%)和 用户代理服务器(uas). SIP的终端可建立语音通话直接而不需要任何其他元素. 图6显示了一个例子,其中user1user2电话发送邀请请求原始含有user1支持能力接收音频和一个UDP端口(port12345andmlawcodec). 当user2inivite接到请求, 他可以建立一个语音频道12345udp港口user1虽然他接受请求,派遣一个 ok回应讯息. 此外,user2反应包括自身媒体潜力, 所用user1建立语音频道(GSM的codec在54321港口为例),并派 一个ack消息承认user2的回应. 为了终止连接,任何一方可以将轮空讯息 必须承认一个OK响应.