SIPintro

SIP Introduction

SIP Components

Server Controls Service Logic

SIP Commands

SIP Headers

Messages 常用代碼

SIP Applications

SIP Applications

SIP Applications

SIP Applications

SIP Phone

Baseline Architecture

Baseline Architecture

Architecture Overview

Baseline Architecture Components

SIP 101

Operation in Proxy Mode

Operation in Redirect Mode

User Registration

SIP Protocol Stack

Application Servers

Application Servers Connectivity

Application Server Platform

Possible AP Server Applications

AP Server Execution

SIPService Creation

SIP Program Example

SIP Program Example

XML Style CPL

RFC List and FAQ

Document Tree

Untitled Document

SIP Introduction

IETF RFC 2543

詳細結構

⊕ SIP Benefits

Open, simple, extensible, and lightweight protocol
Design for IPNeetworks – easier to integrate with telephony and Internet functions
Supports multiple call legs (i.e., forking)
Same protocol used between services and call control entities
Text-based encoding

⊕ SIP Shortcoming

SIP's current features are mostly for call initiation

Need features for post-call setup

Transfer Multi-party Dial-in bridges
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Open, simple, extensible, and lightweight protocol
Design for IPNeetworks – easier to integrate with telephony and Internet functions
Supports multiple call legs (i.e., forking)
Same protocol used between services and call control entities
Text-based encoding

Untitled Document

SIP Components

詳細結構

⊕ User Agents

User Agent Client - UAC	Initiate SIP Requests
User Agent Server - UAS	Accepts or Rejects call

Reside in	Softswitches	IP and soft phones
Handheld and wireless devices	DSL/Cable equipment	PBX/UnPBX

⊕ Servers

Proxy	Heart of SIP network which contains all service logic
Redirect	Returns routing information to the initiating endpoint
Registration	Registration enables subscriber mobility

Reside in

Service Provider Networks	ITSP's Long distance carriers Cable MSO’s
Firewalls	Network address translations
Network Access Points	Cross network interoperability

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Server Controls Service Logic

Server is the Primary Place for SIP Services

Controls all service logic

Central point for location and billing services

Internet Integration

SIP integrates with web, email and chat applications

Call Processing Language

XML language for creating services and applications

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SIP Commands

Invite	主要為啟發一多媒體服務，內含主叫方及被叫方的資訊、和所要進行的服務類別等訊息。
Ack	當一服務要求送達UAC後,需由UAC回應一訊息至UAS端，表示己收到Invite 訊息。
Options	此一訊令主要是用來了解User agent 所能處理事務有那些。如可處理Media type 。
BYE	結束一多媒體服務時所使用的指令。
Cancel	由主叫方取消其服務要求時所使用之指令。
Register	由User agent主動向Registrar Server報告目前位置，供未來各類服務使用。

Tue Apr 26 09:03:33 CST 2011 Untitled Document

SIP Headers

General Headers

Call-ID	Contact	CSeq	Date	Encryption	Expires
From	Record-Route	Timestamp	To	Via

Entity Headers

Content-Encoding	Content-Length	Content-Type

Request Headers

Accept	Accept-Encoding	Accept-Language	Authorization	Contact
Max-Forwards	Organization	Priority	Proxy-Authorization	Proxy-Require
Route	Require	Response-Key	Subject	User-Agent

Response Headers

Allow	Proxy-Authenticate	Proxy-After	Server
Unsupported	Warning	WWW-Authenticate

Tue Apr 26 09:03:33 CST 2011 Untitled Document

Messages 常用代碼

1xx		Provisional
100	Trying	180	Ringing	181	Call Is Being Forwarded	182	Queued
2xx		Successful
200	OK
3xx		Redirection
300	Multiple Choices	301	Moved Permanently	302	Moved Temporarily	305	User Proxy
380	Alternative Service
4xx		Failure
400	Bad Request	401	Unauthorized	402	Payment Required	403	Forbidden
404	Not Found	405	Method Not Allowed	409	Conflict	410	Gone
411	Length Required	413	Request Entity Too Large	414	Request URL Too Long	415	Unsupported Media Type
420	Bad Extension	480	Temporarily Unavailable	481	Call Leg Transaction Does Not Exist	482	Loop Detected
483	Too Many Hops	484	Address Incomplete	485	Ambiguous	486	Busy Here
5xx		Server Failure
500	Server Internal Error	501	Not Implemented	502	Bad Gateway	503	Service Unavailable
504	Gateway Time-out	505	Version Not Supported
6xx		Global Failure
600	Busy Everywhere	603	Decline	604	Does Not Exist Anywhere	606	Not Acceptable

Tue Apr 26 09:03:33 CST 2011 Untitled Document

SIP Applications

Integrate Telephony Services with

web	email	text chat
instant messging	interactive games

Examples

IM Notify when busy	subscriber gets instant messages when friends phones (IP or POTS) available
Call redirect to web	web page returnd instead of busy signal
Web IVR	web page of menus, final choice rings phone
Shared web browsing	talk and browse jointly
Transfer to email	Caller is disconnected and mail tool pops up
Email call logs	Unanswered calls cause email notification
IM notification of conference join	On a conference bridge, instant message indicates participant joins/leaves
Web call-ID	web page of caller pops up when phone rings

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SIP Applications

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SIP Applications

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SIP Applications

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SIP Phone

MIC Worldcom
3Com
Nortel
Siemens
Cisco
PingTel

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Baseline Architecture

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Baseline Architecture

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Architecture Overview

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Baseline Architecture Components

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SIP 101

1	SIP = signaling protocol for establiing session/calls/conference/...
2	session = audio, video, games, chat, ...
3	called server may map name to user@host
4	callee acceps, rejects, forward to new address
5	if new address, go to step 2
7	conversation ...
8	caller or callee sends BYE

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Operation in Proxy Mode

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Operation in Redirect Mode

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User Registration

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SIP Protocol Stack

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Application Servers

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Application Servers Connectivity

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Application Server Platform

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Possible AP Server Applications

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AP Server Execution

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SIPService Creation

Easy service creation based on open standards by

Service providers,

End users,

3rd parties

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SIP Program Example


<call>
 <location url="sip:jones@pc.ex.com">
  <proxy timeout="8s">
   <busy>
    <location url="sip:jones@vmail.ex.com”
     merge="clear” id="voicemail" >
     <proxy />
    </location>
   </busy>
   <noanswer>
    <link ref="voicemail" />
   </noanswer>
  </proxy>
 </location>
</call>

SIP Program Example

INVITE sip:jdrosen@bell-labs.com SIP/2.0
To: sip:jdrosen@bell-labs.com
From: sip:machine@bell-labs.com
Call-ID: 10
Cseq: 0 INVITE
Content-Length: 0
PROXY_REQUEST_TO sip:hgs@cs.columbia.edu SIP/2.0
Max-Forwards:
SIP/2.0 180 Ringing User
CGI_SCRIPT_COOKIE aoi988ans0naa SIP/2.0

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SIP Program Example


<call>
 <location url="sip:jones@pc.ex.com">
  <proxy timeout="8s">
   <busy>
    <location url="sip:jones@vmail.ex.com”
     merge="clear” id="voicemail" >
     <proxy />
    </location>
   </busy>
   <noanswer>
    <link ref="voicemail" />
   </noanswer>
  </proxy>
 </location>
</call>

SIP Program Example

INVITE sip:jdrosen@bell-labs.com SIP/2.0
To: sip:jdrosen@bell-labs.com
From: sip:machine@bell-labs.com
Call-ID: 10
Cseq: 0 INVITE
Content-Length: 0
PROXY_REQUEST_TO sip:hgs@cs.columbia.edu SIP/2.0
Max-Forwards:
SIP/2.0 180 Ringing User
CGI_SCRIPT_COOKIE aoi988ans0naa SIP/2.0

Tue Apr 26 09:03:39 CST 2011 Untitled Document

XML Style CPL

<call>
 <string-switch field=“from”>
  <string is=“boss@company.com”>
   <location url=“sip:joe@att.com”>
    <proxy>
     <busy>
      <location url=“tel:5551212”>
       <proxy>
        <busy>
         <location url=“sip:voicemail@att.com”
                   link=“vm”>
          <proxy/>
         </location>
        </busy>
        <noanswer>
         <link id=“vm”/>
        </noanswer>
       </proxy>
      </location>
     </busy>
     <noanswer>
      <link id=“vm”/>
     </noanswer>
    </proxy>
   </location>
  </string>
  <otherwise>
   <link id=“vm”/>
  </otherwise>
 </string-switch>
</call>

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RFC List and FAQ

FAQ

RFC 1806

RFC 2068

RFC 2234

RFC 2543

RFC 2806

Tue Apr 26 09:03:39 CST 2011 Untitled Document

Document Tree

SIP Architecture and Functionality

Guidelines for Authors of SIP Extensions
SIP MIB
SIP and SOAP
SIP Extensions for supporting distributed call state
SIP INFO vn. 5
SIP INFO method for event reporting
SIP INFO method for DTMF digit transport and collection
SDP media alignment in SIP

Services

Emergency Call Services (911)
A SPIRITS solution based on virtual SIP user agents
Third party call control in SIP
SIP message waiting
SIP call control transfer
SIP for the hearing disabled
SIP for home appliances

Infrastructure: AAA, QoS and Security

SIP transport of OSP token
SIP firewall solution

PSTN and H.323 support

H.323-SIP
MIME media types for ISUP and QSIG objects

Tue Apr 26 09:03:40 CST 2011