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					   IETF VoIP 표준 기술




        최선완
   안양대학교 정보통신공학과
    sip: sunchoi@anyang.ac.kr
mailto:sunchoi@aycc.anyang.ac.kr
Next Generation Network
 Network convergence
   Traditional Telephone Networks
   Internet Telephony
 Service convergence
   Traditional separated service for voice and data
   Integrated services for voice, data, and video
 Two dimensions for converged network/service
   Openness
       - API (Application Programming Interfaces)
       - User’s application & service creation capabilities
   Web-based provisioning
 Softswitch
 Next Generation Network Architecture
                                   Service         GUI
                                definition and
                                  execution        Java
                                                            Customer
- Service definition                                      Service provider
- Billing                               Service
- Provisioning       Service            programs
                     Applets                              SCP
                                                                TCAP/SS7
    Customer                                                               Public
    Care and                                           SS7
      billing                   Softswitch           Gateway
                                                                     Signaling
                                                           TCAP/SS7
   Network                                                  ISUP/SS7
    OSSs                                                             Network
                      Announcement      Feature
                         server          server

  Residential        Voice/IP    Voice/IP/ATM         Trunking
   Gateway                          SONET             Gateway        PSTN
                                  Backbone
                                   network
           Next-gen architecture:
           Access  Applications infrastructure
            Access/Transport                 Apps/Services
                                               App/Feature
                                            App/Feature
                                                   Server
                                         App/Feature
                                                Server
                               SIP     App/Feature
               Soft Switch                  Server
                                         Server
              MGCP/MEGACO            SIP &                     Control
                                     VXML
 PSTN                                                           Media

                 Media         RTP        Media
 Cable
                Gateway                   Server
                                                             Disk
Wireless
SYNDEO Class 5 IP Softswitch
미디어 서버 기능
 응용     Conference                      Announcement
                            UMS            Server
 서버       Bridge
      Prompt Library   Prompt Library   Prompt Library
 미    DTMF Collect     DTMF Collect     Play Only
 디    Play-Mix-        Play-Record      G.711 Only
 어    Record           Web Content      Low Usage
 서    Multi-Codec      Email Fetch
 버    On-demand        Fax Retrieval
      Dynamic Usage    Multi-codec
                       Sporadic Usage
IETF 표준화 회의

연 3회 (1회는 미국 아닌 곳)
Future Meeting
 50th, Minneapolis, 2001. 3 .18 – 3. 23
 51th, London, 8/5-8/10
 52th, Salt Lake City, 12/9-12/14
 53th, Minneapolis, 2002. 3. 17 -22
 54th, Yokohama, July 2002
 55th, Atlanta, Nov. 2002.
 56th, San Francisco, Mar. 2003.
 59th, Seoul, Mar. 2004.
IETF VoIP Related WG
 Transport Area                              Application Area
       Audio/Video Transport (avt)               SIP for Instant Messaging and
       Telephone Number Mapping (enum)            Presence Leveraging Extensions
       IP Telephony (iptel)                       (simple)
       Media Gateway Control (megaco)            Instant Messaging and Presence
       Middlebox Communication (midcom)           Protocol (impp)
       Multiparty Multimedia Session
        Control (mmusic)
       Signaling Transport (sigtran)
       Session Initiation Protocol (sip)
       Session Initiation Proposal
        InvestiGation (sipping)
       Service in the PSTN/IN Requesting
        InTernet Service (spirits)
       Speech Services Control (speechsc)
       PINT (PSTN-Internet Interworking)
        WG (closed)
표준화 현황 – IPTEL WG
   개요
      VoIP 서버에서 Call을 처리할 수 있는 기                                       Intradomain
                                                                          Protocol
     술 표준화                                                                  (SLP)
    VoIP 네트워크에서 VoIP Gateway용 라
     우팅 프로토콜 표준화
 표준화 추진 현황
     RFC
          RFC 2874, Call Processing Language
           Framework and Requirements, 2000.5.
          RFC 2871, A Framework for Telephony
           Routing over IP, 2000. 6.                         Location        TRIP
          RFC 3219, Telephony Routing over IP                Server
           (TRIP), Jan. 2002.
     Internet Draft
                                                      User                            Front-
          Cpl-06 (CPL : A Language for User         Agent
           Control of Internet Telephony Services)                                      End
          TRIP-MIB-03 (Management Information
           Base for Telephony Routing over IP                             Gateway
           (TRIP))
          tgrep-00 (A Telephony Gateway
           REgistration Protocol (TGREP))
Sample CPL Script: Graphical
Version
             mandatory

    Address-switch       location                                  busy
      field:                                      proxy           timeout
                           url: sip:jychun@
                                                     timeout: 10s failure
    origin                            kbs.co.kr

     subfield:host
     Address
       subdomain-             Voicemail
Call of:                      location
                                url: sip:hksohng@
    kbs.co.kr                         voicemail.kbs.co,kr      redirect
    otherwise
Sample CPL Scrip: XML Version
<?xml version="1.0" ?>
  <!DOCTYPE cpl PUBLIC "-//IETF//DTD RFCxxxx CPL 1.0//EN" "cpl.dtd">

 <cpl>
  <subaction id="voicemail">
   <location url="sip:hksohng@voicemail.kbs.co.kr">
     <redirect />
   </location>
  </subaction>

  <incoming>
   <address-switch field="origin" subfield="host">
     <address subdomain-of=“kbs.co.kr">
      <location url="sip:hksohng@kbs.co.kr">
        <proxy timeout="10">
         <busy> <sub ref="voicemail" /> </busy>
         <noanswer> <sub ref="voicemail" /> </noanswer>
         <failure> <sub ref="voicemail" /> </failure>
        </proxy>
      </location>
     </address>
     <otherwise>
      <sub ref="voicemail" />
     </otherwise>
   </address-switch>
  </incoming>
 </cpl>
Generative of CPL


        CPL Generator
        에 의한 XML
         문서의 생성
Server’s CPL Processing
  Call 이 SIP Server에 수신되었을 때 call에 대한 정보를 확인하여,
   call에 대한 CPL script 를 수행
                                 Service
                                              user b-1 setting
                                  logic       if busy, to user b-2

            4.user b-1 is busy             5. Call to User b-2


          1.Call to user b-1                   2.Call
 User a                           SIP                            User b-1
                                               3.Busy
                                 Server
                                                    6.call
                                                                 User b-2
                         7.communication                     Connection
표준화 현황 – MEGACO WG
 개요
     다양한 VoIP용 media gateway(MG)를 제
       어하는 프로토콜 표준화
                                                        Context
   표준화 현황                                                                                       Termination
                                                                                              SCN Bearer Channel
      RFC
          RFC 2805, Media Gateway Control
           Protocol Architecture and Requirements,
                                                         Termination
           2000. 4.                                                                              Termination
                                                         RTP Stream
          RFC 2885, Megaco Protocol version 0.8,                                             SCN Bearer Channel
           2000. 8.
          RFC 2886, Megaco Errata, 2000. 8.
          RFC 3015, Megaco Protocol Version 1.0,
           2000.11
                                                                                    Context
          RFC 3054, Megaco IP Phone Media              Termination                              Termination
           Gateway Application Profile, 2001. 1.     SCN Bearer Channel                       SCN Bearer Channel

     Internet Draft
          Megaco/H.248v2
          mib-04 (Megaco MIB)
          naspkg-04 (Megaco/H.248 NAS Packages)                          Context
          3015corr-02 (Megaco Protocol Version 1         Termination                            Termination
           With Corrections)                              RTP Stream                          SCN Bearer Channel
          h248v2-02 (The Megaco/H.248 Gateway
           Control Protocol, version 2)
          callflows-00 (Megaco/H.248 Call flow
           examples )
Media Gateway Control Protocol -
MGCP

RFC 2705
Controlling VoIP Gateways from External Call
 Control Elements
History
   Simple Gateway Control Protocol (SGCP): Bellcore/Cisco
   IP Device Control (IPDC): Level 3 TAC extended work of XCOM,
     Ascend & others
   Media Gateway Control Protocol (MGCP): Merged IPDC + SGCP
   Media Device Control Protocol (MDCP): Lucent
   H.GCP: ITU SG16 Functional Decomposition Control Protocol
   MEGACO: IETF
The MGCP ecosystem
 Media stream                                            optional

                       Call Agent                     Signaling
                                                        Entity
  MGCP/UDP                                      MGCP/UDP


                          RTP/UDP
       Gateway                                 Gateway
                          Or AAL2

                      IP Network or
                          ATM
                Switched circuit network (or
                    other technologies)
  Key elements of the protocol
                       . Local Name
        Call Agent     . Domain Name

                      . Transaction Id
                      . Transaction Verbe
MGCP grammar uses a
                      . Response Code
BNF text format.
                      . Connect Id
                      . Session Description Protocol


        Gateway        . Endpoint Name
                       . Domain Name
Commands
Each command uses a transaction
Call Agent to Gateway :
  EPCF : EndpointConfiguration
  RQNT : NotificationRequest
  CRCX : CreateConnection
  MDCX : ModifyConnection
  DLCX : DeleteConnection
  AUEP : AuditEndpoint
  AUCX : AuditConnection


Gateway to Call Agent :
  NTFY : Notify
  RSIP : RestartInProgress
표준화 현황 – MEGACO WG
 개요
     다양한 VoIP용 media gateway(MG)를 제
       어하는 프로토콜 표준화
                                                        Context
   표준화 현황                                                                                       Termination
                                                                                              SCN Bearer Channel
      RFC
          RFC 2805, Media Gateway Control
           Protocol Architecture and Requirements,
                                                         Termination
           2000. 4.                                                                              Termination
                                                         RTP Stream
          RFC 2885, Megaco Protocol version 0.8,                                             SCN Bearer Channel
           2000. 8.
          RFC 2886, Megaco Errata, 2000. 8.
          RFC 3015, Megaco Protocol Version 1.0,
           2000.11
                                                                                    Context
          RFC 3054, Megaco IP Phone Media              Termination                              Termination
           Gateway Application Profile, 2001. 1.     SCN Bearer Channel                       SCN Bearer Channel

     Internet Draft
          Megaco/H.248v2
          mib-04 (Megaco MIB)
          naspkg-04 (Megaco/H.248 NAS Packages)                          Context
          3015corr-02 (Megaco Protocol Version 1         Termination                            Termination
           With Corrections)                              RTP Stream                          SCN Bearer Channel
          h248v2-02 (The Megaco/H.248 Gateway
           Control Protocol, version 2)
          callflows-00 (Megaco/H.248 Call flow
           examples )
Commands

 Add
   Adds a Termination to a Context
 Subtract
   Removes a Termination from a Context
 Move
   Moves a Termination from one Context to
    another
Commands

 Modify
   Changes Properties of a Termination
 AuditCapabilities & AuditValue
   Discovers what Terminations are realized &
    what capabilities they have
 Notify
   Notifies MGC of events occuring on
    Terminations in the MG
Packages
   Grouping of termination characteristics
   Properties - parameters on a Termination
    that can be set or examined
   Events - detected state/status change on a
    Termination
   Signals – tones, announcements, etc.
    applied to a Termination
   Statistics - packet/cell counts, errors, etc.
Events

  Asynchronous events detected on a
   termination
  Named, defined in a Package
  MGC sets a list of events to detect on a
   Termination
  MG sends Notify command when detected
  RequestID correlates request with Notify
표준화 현황 - PINT
              INVITE                                                                      SIP
                                                                           IP Network
                       IP Connection                                                    SIP
                                                               Gateway


                       PSTN/IN                     Service
                                                   Node (SN)     Service
                                                                 Control
                                                                 Point (SCP)
                                CO

                          MSC
                                       Mobile
                                       Switching
                                       Center


                                                                    Request-to-FAX
 Request-to-Dial
                                                                    Completion
 Completion

                                                                    J. Kojik, VoN2000
표준화 현황 – SIGTRAN WG
   개요                                                                             SCN Protocols
      Signaling Gateway 프로토콜 표준화
   표준화 현황
                                                                              SCN Adaptation Module
      RFC
          RFC 2719, Framework Architecture for Signaling Transport,
           1999. 10.
          RFC 2960, Stream Control Transmission Protocol, 2000. 10.      Common Signaling Transport
          RFC 3057, ISDN Q.921-User Adaptation Layer, 2001. 2.
          RFC 3257, Stream Control Transmission Protocol Applicability
           Statement. 2002. 4.                                                  Standard IP Transport
          RFC 3331, Signaling System 7 (SS7) Message Transfer Part 2
           (MTP2) - User Adaptation Layer, 2002. 9.
          RFC 3332, Signaling System 7 (SS7) Message Transfer Part 3                    IP
           (MTP3) - User Adaptation Layer (M3UA ), 2002. 9.
      WGLC
          Signaling-over-SCTP-Applic-05 (Telephony Signaling Transport
           over SCTP Applicability Statement)
          SUA-14(SS7 SCCP-User Adaptation Layer)
          M2PA-06(SS7 MTP2-User Peer-to-Peer Adaptation Layer)
          DUA-03 (DPNSS/DASS 2 extensions to the IUA protocol )
          SCTP-MIB-07
          M3UA-MIB-04
          V5UA-03 (V5.2-User Adaptation Layer (V5UA))
          m3ua-implementors-guide-01 (M3UA ImplementorÆs Guide)
          iua-imp-guide-01 (IUA (RFC 3057) Outstanding Issues)
표준화 현황 – SPIRITS WG
 개요
    IN(지능망) 또는 PSTN에서 인                                                 PSTN/IN                                             Internet
     터넷에 요청한 서비스를 지원
     하기 위한 표준화                                                      Proprietary
 표준화 현황                                                   (ICW SL) Interface UAC/UAS
                                                                                                        PINT
                                                                                                                          (Proxy Server)
                                                                                                                                                PINT
                                                           SCF/SDF              SCGF                                    ICW Server System
    RFC                                                                                                Firewall
                                                                           SCP
            RFC 2995, Pre-SPIRITS                                                                                                                 Firewall
             Implementations of PSTN-initiated             INAP                                                 (Callee)
                                                                                    INAP
             Services, 2000.11                                                                              ICW Subscribe’s
            RFC 3136, The SPIRITS Architecture                IP                        SSP                  Phone and PC
            RFC 3298, SPIRITS Protocol
             Requirements, 2002. 8.                                                                                                 (UAC/UAS)
                                                                Caller                   LEX                                      ICW Client System
 Internet Draft
     Protocol-02 (The SPIRITS Protocol)
     in-03 (On selection of IN parameters to be carried   INAP: Intelligent Network Application Protocol          SCF: Service Control Function
      by the SPIRITS Protocol)                             PINT: PSTN/Internet Interworking Protocol               SDF: Service Data Function
                                                           SL: Service Logic
     sip-evt-package-02 (Toward the Definition of the     UAS: User Agent Server
                                                                                                                   SCGF: Service Control Gateway Function
                                                                                                                   SCP: Service Control Point
      SIP Events Package for SPIRITS Protocol)             UAC: User Agent Client                                  SSP: Service Switching Point
     mobility-00 (Mobility Events Management in
      SPIRITS)
표준화 현황 - ENUM
 개요
    전화번호를 그 전화번호와 관                                        NAPTR Resource Record
     련된 자원과 접촉하기 위한                                         SRV RR
     attributes들(예, URLs)로
     mapping하기 위해서 DNS를 기
     반으로 한 구조 및 프로토콜 정                                                  IP
     의                                                               NETWORK


 표준화 현황
    RFC 2916, E.164 number and
     DNS                                                                 GITD
    Internet Draft
        e164-gstn-np-05 (Number Portability in the
         GSTN: An Overview)
        rfc2916bis-01 (The E.164 to URI DDDS
         Application)
        epp-e164-01 (Extensible Provisioning
         Protocol E.164 Number Mapping)
        usage-scenarios-00 (ENUM Usage                                 TELEPHON
                                                    Enterprise/ SP                 Enterprise/ SP #2
         Scenarios)                                 #1
                                                                               E
                                                                        NETWORK
Brief History of ENUM

IETF E.164-to-IP BOF on mapping phone
 numbers to IP address – 8/98
IETF formed tElephone NUMber mapping
 (enum) WG in late 1999
RFC2916 published in 9/2000
IETF Proposed Standard
Simple Description


E.164 Number               URI
+1-919-555-1212   ENUM     sip:foo@bar.com



                         IP 주소 =
URI               DNS    203.232.130.56
sip:foo@bar.com   SRV    Port = 5060
                         Protocol = UDP
                         Weight = 3
                         Priority = 1
The “common” ENUM domain e164.arpa
                                      The root node
                                            “.”

                                              com                                   net
        arpa                                                      ...


                 in-addr.arpa           second-level node       second-level node         second-level node
e164.arpa

                               third-level node     third-level node
    1.e164.arpa


        4.1.e164.arpa

                         ...


            1.2.3.4.5.5.5.2.4.0.4.1.e164.arpa
Simple Example
                                      DNS-Server
             Query                  Response
 4.3.2.1.7.9.8.6.4.e164.arpa?       IN NAPTR 10 10 “U” “sip+E2U”“!^.*$! sip:spam@paf.se!”




                                                       “Invite”


           Dial                           Sip
        +468971234               Invite:spam@paf.se


                     Sip proxy                           Sip proxy
Structure of E.164 Number
Structure to use for geographic areas
        CC            NDC                       SN

     1-3 digits     N digits             Max (15-N) digits


                            National (significant) number


             International public telecommunication
                 Number for geographical areas

       CC – Country Code
       NDC – National Destination Code
       SN – Subscriber Number
  Delegations
                        .                     ns.nic-se.se.
                                              ns.se.     IN NS ns.nic-se.se.
 Root server .                          se.
 se.           IN NS ns.nic-se.se.            a.se.      IN NS ns.a.se.
 ns.nic-se.se. IN A 192.168.0.1               c.se.      IN NS ns.c.se.
                                              ns.nic-se.se. IN A 192.168.0.1
                                              ns.a.se.    IN A 192.168.1.1
               c.se.                          ns.c.se.    IN A 192.168.3.1
                                     a.se.
                                              ns.a.se.
ns.c.se.                                      a.se.        IN   NS    ns.a.se.
c.se.    IN NS ns.c.se.                       b.a.se.      IN   NS    ns.b.a.se.
ns.c.se. IN A 192.168.3.1                     ns.a.se.     IN   A    192.168.1.1
                                              ns.b.a.se.   IN   A    192.168.2.1
                                        b.a.se.

                                              ns.b.a.se.
                                              b.a.se.    IN NS ns.b.a.se.
                                              ns.b.a.se. IN A 192.168.3.1
   Example DNS Entry
$ORIGIN 4.3.2.1.7.9.8.6.4.e164.arpa.
@ IN NAPTR 10 10 "U" "sip+E2U" \
      "!^.*$!sip:spam@paf.se!"
@ IN NAPTR 20 10 "U" "mailto+E2U" \
      "!^.*$!mailto:spam@paf.se! "
@ IN NAPTR 30 10 "U" "ldap+E2U" "!^+46(.*)$!ldap://ldap.telco.se/cn=0\1"
SIP: Locating SIP Servers - DNS

Load balancing and fail
 over
  DNS NAPTR
  DNS SRV
Transport selection
DNS (Domain Name System)
 SRV RR(Resource Record)
    RFC 2782, A DNS RR for specifying the location of services (DNS SRV)
    Format
   _Service._Proto.Name TTL Class SRV Priority Weight Port Target
            •   Service: the symbolic name of the desired service
            •   Proto: the symbolic name of the desired protocol
            •   Name: the domain this RR refers to.
            •   TTL: time to live of the RR => cache
            •   Class: IN(Internet Family), CH(Chaos Family)
            •   Priority: the priority of the target host
                  – client must attempt to contact the target host with the
                    lowest-number priority
            • Weight: server selection mechanism (within same priority)
            • Port
            • Target: the domain name of the target
SRV RR
   $ORIGIN example.com.
    @         SOA server.example.com. root.example.com. (
         1995032001 3600 3600 604800 86400 )
         NS server.example.com.
         NS ns1.ip-provider.net.
         NS ns2.ip-provider.net.                              Priority
   ; foobar - use old-slow-box or new-fast-box if either is
    ; available, make three quarters of the logins go to      Weight
    ; new-fast-box.
    _sip._tcp SRV 0 1 5060 old-slow-box.example.com.
         SRV 0 3 5060 new-fast-box.example.com.
   ; if neither old-slow-box or new-fast-box is up, switch to
    ; using the sysdmin's box and the server

        SRV 1 0 5060 sysadmins-box.example.com.
        SRV 1 0 5060 server.example.com.
   server       A 172.30.79.10
    old-slow-box           A 172.30.79.11
   sysadmins-box           A 172.30.79.12
   new-fast-box            A 172.30.79.13
    ; NO other services are supported
   *._tcp SRV 0 0 0 .
   *._udp SRV 0 0 0 .
표준화 현황 –MMUSIC WG
                                                           Internet Draft
 개요                                                           sdp-new-10
    멀티미디어 회의용 표준                                                   SDP with SIP
                                                                    ..
 RFC                                                          sdpng-05 (Session Description and
     RFC 2326, Real Time Streaming Protocol (RTSP),            Capability Negotiation)
      1998. 4.                                                 srcfilter-01 (SDP Source-Filters)
     RFC 2327, SDP: Session Description Protocol,             fid-06 (Grouping of media lines in SDP )
      April 1998.
     RFC 2543, SIP: Session Initiation Protocol, 1999.
                                                               sdp-comedia-04 (Connection-Oriented
      3.                                                        Media Transport in SDP)
     RFC 2974, SAP: Session Announcement Protocol,            sdpng-trans-01 (SDPng Transition )
      2000. 10.                                                natreq4udp-00 (Short term NAT
     RFC 3108, Conventions for the use of the Session          requirements for UDP based peer-to-peer
      Description Protocol (SDP)       for ATM Bearer
      Connections, 2001. 5.                                     applications)
     RFC 3259, A Message Bus for Local                        sdp4nat-02 (RTCP attribute in SDP),
      Coordination, 2002. 4.                                    expires
     RFC 3264, An Offer/Answer Model with the                 kmgmt-ext-05 (Key Management
      Session Description Protocol (SDP), 2002. 6.              Extensions for SDP and RTSP )
     RFC 3266, Support for IPv6 in Session                    rfc2326bis-01 (Real Time Streaming
      Description Protocol (SDP), 2002. 6.
                                                                Protocol (RTSP) )
SDP [RFC 2327]

Session Information Description Format ?
to convey information about media streams in
 multimedia sessions, to allow the recipients of a
 session description to participate in the session
in multicast based sessions on the internet
   a mean to communicate the existence of a session
   a mean to convey sufficient information to enable joining
     and participating in the session
What is Session ?

A multimedia session
  A set of multimedia senders and receivers and the data
   streams flowing from senders and receivers
A multimedia conference
  An example of a multimedia session
A session
  Can comprise one or more RTP sessions
  Origin field in SDP
     <user name, session id, network type, address type, address>
SDP SPECIFICATION (I)

Descriptions are Textual: <type>=<value>
                    v= (protocol version)
                       o= (owner/creator and session
  Type == smalll           identifier)
                       s= (session name)
Consists of           i=* (session information)
                       u=* (URL of description)            *: options
  Session description e=* (email address)
                       p=* (phone number)
  Time description c= (connection information)
                       b=* (bandwidth information)
  Media description        One or more time descriptions
                       k=* (encryption key)
Session description   a=* (zero or more session attribute
                            lines)
                           Zero or more media descriptions
SDP SPECIFICATION (II)

Time description session is active)
         t = (time the
            r =* (zero or more repeat times)




Media description and transport address)
         m= (media name
            i=* (media title)
            c=* (connection information – optional if
                 included at session-level)
            b=* (bandwidth information)
            k=* (encryption key)
            a=* (zero or more media attribute lines)
SDP DESCRIPTION – EXAMPLE
(1/2)
Address id Session-id Version Network Address Address
     User
     (login)                     type      type
  v= 0                                                  Global
  o= sunchoi 2890844526 2890842087 IN IP4 126.16.64.4   session
  s= SDP Seminar                                        uniquie
  I= A Seminar on the Session Description Protocol      id
  u= http://aycc.anynag.ac.kr/presentation/sdp.pdf
  e= sunchoi@aycc.anyang.ac.kr (Sunwan Choi)
                                                        TTL (if IPv4 & multicast)
  p= +82-31-467-0884
  c= IN IP4 224.2.17.12/127                              NTP timestamp
  b= CT:64
  t= 2873397496 2873404696
  r= 604800 3600 0 90000                     <1주일, 1시간, 지정시간, 다음시간>
  a= recvonly                  G.723
  m=audio 3456 RTP/AVT 4 18          G.729
  a= rtpmap:4 G723/8000
  a= rtpmap:18 G729/8000
  m=video 2232 RTP H261
  m=whiteboard 32146 UDP WB
  a= orient:landscape
SDP DESCRIPTION – EXAMPLE
(2/2)
                 Session-id Version
Example
    v= 0
    o=   sunchoi 2890844526 2890842087 IN IP4 126.16.64.4
    s=   INFOCOM’99 NGI TUTORIAL
    I=   A Turorial on IETF IP Telephony
    u=   http://ice.anynag.ac.kr/presentation/ngi99.ps
    e=   sunchoi@aycc.anyang.ac.kr
    c=   IN IP4 224.2.17.12/127                        Conn. info.
    b=   CT:64
    t=                                                 NTP
         2873397496 0
    a=                                                 timestamp
         recvonly
    m=   audio 3456 RTP/AVP 96
    m=   video 2232 RTP H261 31
    m=   whiteboard 32146 UDP WB
    a=   orient:landscape
표준화 현황 - MIDCOM WG

Comm. With NAT and                           External     Server               Internal
                                               Host INVITE                       Server
 Firewall
                                                    Response
RFC
     RFC 3303, Middlebox communication
      (midcom) architecture and framework
     RFC 3304, Middlebox Communications
      (midcom) Protocol Requirements        Media over UDP          Firewall/
 Internet Draft                                                      NAT
     Stun-02 (STUN - Simple Traversal of
      UDP Through Network Address                                                   Internal
                                                                                      Host
      Translators )
     Protocol-eval-05 (Middlebox
      Communications (MIDCOM) Protocol
      Evaluation)
표준화 현황 –IMPP & SIMPLE
WG
 IMPP WG
                                                        SIMPLE WG
    IMPP 프레임워크 표준
   RFC                                                   SIP 기반 IMPP 표준 프로토
         RFC 2778, A Model for Presence                    콜
          and Instant Messaging, 2000. 2.                 presence-07 (SIP Extensions for
         RFC 2779, Instant Messaging /                    Presence),
          Presence Protocol Requirements,
          2000. 2.                                        winfo-package-02 (A SIP Event Sub-
           RFC 3339, Date and Time on the Internet:       Package for Watcher Information)
            Timestamps                                    winfo-format-02 (An XML Based
     Internet Draft                                       Format for Watcher Information)
         cpim-03 (Common Presence and                    cpim-mapping-01 (CPIM Mapping of
            Instant Messaging)                             SIMPLE Presence and Instant Messaging)
         msgfmt-06 (Common Presence and                  presencelist-package-00 (A SIP Event
            Instant Messaging: Message Format)
                                                           Package for List Presence)
         pidf-05 (CPIM Presence Information
            Data Format)
SIMPLE MODEL
                                          Presence
                                           Server
MESSAGE                                   (optional)


SUBSCRIBE/NOTIFY

                       Presenc                    Proxy                  Presenc
                       e                         Server                  e
                       Clients                  (optional)               Clients

          Presence
          Server

WATCHER              PRESENTIT
                     Y
                                      Instant        Message
                                      Service



                             Sender                            Instant
                                                               InBox
Windows Messenger




        4255551212
Session Initiation Protocol
(SIP)
SIP & SIPPING WG
              Request
                                                SIP Redirect
              Response                             Server
                                                                     Location Service




                               2                                         Location
                                                                          Server
                                   3

                                                        5
                               4
                                                            6
                      1                11
                                                                 7
                        12                                  10                SIP Proxy
                                            SIP Proxy


                                                                     8
          SIP Client
                                                                     9
     (UAC:User Agent Client)

                                                  SIP Client
                                              (User Agent Server)
SIP Timeline
 1996                            1999. 9.
   Mark Hadley’s SIP(Session       IETF SIP WG
     Invitation Protocol)         2000. 6
   Henning Schulzrinne’s           RFC 2543bis-01
     SCIP(Simple Conference       2000. 9
     Control Protocol)
                                    RFC 2543bis-2 (Draft)
 1996-1997
                                  2001.3
   Interest grows in academic
     circles over MBone             Split into SIP WG &
                                      SIPPING WG
 1998
                                    SIMPLE WG (Application
   MCI’s Henry Sinnreich =>          Area)
     VoIP Appliance
                                  2002. 7
 1999. 3.
                                    RFC 3261
   RFC 2543 by IETF MMUSIC
     WG
SIP Adoption of Key Standard
Bodies
 IETF PINT
   http://www.ietf.org/html.charters/pint-charter.html
 3GPP
   http://www.3gpp.org
 Softswitch Consortium
   http://www.softswitch.org
 IMTC & ETSI TIPHON
   http://www.ietf.org & http://www.etsi.org/tiphon/
 PacketCable
   http://www.packetcable.com
 SIP Forum
   http://www.sipforum.org
 Parlay/JAIN
   http://www.parlay.org
   http://java.sun.com/products/jain/index.html
SIP Functions
 IETF-standardized peer-to-peer signaling protocol
 User location
   Locate user given email-style address
   Personal mobility
        Different terminal
        Same identifier
        Any location
 User capabilities
   (re)-negotiate session parameters
 User availability
   Determination of willingness of the called party to engage in
      communications
 Call setup
 Call handling
   Manual and automatic forwarding (“name/number mapping”)
   “forking” of calls
   Terminate and transfer calls
SIP Protocol Overview
 Part of IETF conference control architecture:
    SAP for advertising multimedia sessions
    RTSP for controlling delivery of streaming media [RFC
     2326]
    SDP for describing multimedia sessions [RFC 2327]
    RTP for transporting real-time data and providing QoS
     feedback [RFC 1889]
    RSVP for reserving network resources [RFC 2205]
    others: malloc, multicast, conference bus, ...
 HTTP-like [RFC 2616]
 URI and URL [RFC 2396]
SIP and H.323

       H.323           SIP + SDP
       H.225.0 + RAS   SIP
       H.245           SDP, SMIL
       Gatekeeper      Proxy
SIP Protocol

text based (~ HTTP)
 methods
     INVITE     initiate call
     ACK        confirm final response
     BYE        terminate (and transfer) call
     CANCEL     cancel searches and “ringing”
     OPTIONS    features support by other side
     REGISTER   register with location service
SIP Message


              Syntax: HTTP/1.1과 동일
                        함



   INVITE                            1xx : Informational
   ACK                               2xx : Success
   BYE                               3xx : Redirection
   CANCEL                            4xx : Client Error
   REGISTER                          5xx : Server Error
   OPTION                            6xx : Global Error
SIP Message Format

           start-line

           *message header

           CRLF

           [message body]
SIP Syntax
           request                        response
      method Request-URI SIP/2.0   SIP/2.0 status reason

      Via:              SIP/2.0 protocol host:port
      From:             user <sip:from_user@source>
      To:               user <sip:to_user@destination>
      Call-ID:          localid@host
      CSeq:             seq# method
      Content-Length:   length of body
      Content-Type:     media type of body
      Header:           parameter;par=“value”;par2=“value”;
                        par3=“value folded into next line”
        Blank line
      V= 0
      O= Origin_user timestamp timestamp IN IN4 host
      C= IN IN4 media destination address
      T= 0 0
      M=Media type port RTP/AVP payload types
SIP Architecture

        Request
                                                   SIP Redirect
        Response                                      Server          Location Service

                            2
                                                                          Location
                                    3
                                                                           Server
                                                         5
                                4
                                                             6
                   1                    11
                                                                  7
                       12                                    10
                                             SIP Proxy                         SIP Proxy
                                                                      8
     SIP Client                                                       9
(UAC:User Agent Client)
                                                      SIP Client
                                                  (User Agent Server)
SIP Servers and Clients
 UAC (User-Agent Client)
   caller application
 UAS (User-Agent Server)
   Accept, redirect, refuse call
 Redirect Server
   Redirect requests
 Proxy Server
   Server + client
 Registrar
   Track user locations
 User agent = UAC + UAS
 Often combine registrar + (proxy or redirect server)
SIP Proxy Server
SIP Location Server
DNS & SIP Server
  Forking Proxy Example
                                                              acm.org
                                                                (A)
   Bell (a.g.bell-tel.com) -> Watson(t.watson@ieee.org)
                          sip.iee.org
                                 (P)   2. INVITE (branch=2)
                     1. INVITE                                     Searching
                                                                   …
    c.bell-tel.com
         (C)




                                                              y.bell-tel.com
                                                                   (Y)
h.bell-tel.com
     (H)                                                      x.bell-tel.com
                                                                   (X)
SIP Is ..
 Open Standard
    Download from anywhere                        Application       SIP/XML
    Anyone can implement                          SIP/XML
    Anyone can comment
 Modular                                           Softswitch        SIP-T
    Add without taking away                      MGCP
    Add only what must be added
    Backwards compatible                         Media Gateway      RTP
 Flexible
    Blessing, or Curse is up to us
    Enhancements that make sense
    Logic and features reside in the end-points NOT the protocol.
    Limits on SIP?
 Tool
    Interface for intelligent end points
    No limit on functions
    Facilitate communication between endpoint
      Protocol Support
                                                Application Specific
Application Layer
 • Call intelligence             AS or SCP     Inter-App Protocols      AS or SCP
 • Service creation/ execution
 • Mgmt of provisioning
                                                                                      SIP
                                                                                     TCAP
Call-Control Layer                                 IP: SIP, SIP-T
(Softswitch)                       MGC               Control                   MGC
 • Resource mgmt                                    Protocols
 • Bearer control
 • Call routing/translation                                      M3UA/SCTP
 • IN/AIN                                                          (Sigtran)
                                               VoATM: BICC
                                             TDM: ISUP, Q.931, CAS
                                                                                     MEGACO/H.248
Signaling Layer
 • Signal processing                  SG            Signaling           SG
                                                    Protocols
 • Signal control


Media Layer                                     TDM, IP/RTP, ATM
 • Media processing
 • Media control                    MG                                     MG
                                                Media Transport
 • Tone & announcement
 • QoS
      Layered Control
                                     XML
                      SCE                           Services
                                               Service Repository SIP
                                                   Repository             ASP
                                                   Services
                    SIP                               System Services
                    UA                      SIP                    SIP
                                           ProxyHA                 UA
                    SIP
                    UA


                                      Base       Base     Base
                                      Voice      Data     MM
                                      Serv.       EC
                                                 Serv.    Serv.     SIP
SCE – Service Creation Environment
DCA – Device Control Agent
                                       Media Devices/Hardware
                                              M. Holdrege, SIP 2001, Feb. 2001
Carrier Class SIP Architecture
SIP Services in a 2.5G Network


          Service
          Provider




                           2.5G
                           Wireless
                           Carrier
                           Networks

         2.5G SIP
          Client            출처: VoN
SIP WG Charter
   1. bis: A draft standard version of SIP.
   2. callcontrol: Completion of the SIP call control specifications, which enables multiparty services, such as
    transfer and bridged sessions.
   3. callerpref: Completion of the SIP caller preferences extensions, which enables intelligent call routing
    services.
   4. mib: Define a MIB for SIP nodes.
   5. precon: Completion of the SIP extensions needed to assure satisfaction of external preconditions such as
    QoS establishment.
   6. state: Completion of the SIP extensions needed to manage state within signaling, aka SIP "cookies".
   7. priv: Completion of SIP extensions for security and privacy.
   8. security: Assuring generally adequate security and privacy mechanisms within SIP.
   9. provrel: Completion of the SIP extensions needed for reliability of provisional messages.
   10. servfeat: Completion of the SIP extensions needed for negotiation ofserver features.
   11. sesstimer: Completion of the SIP Session Timer extension.
   12. events: Completion of the SIP Events extensions (Subscribe/Notify).
   13. security: Requirements for Privacy and Security.

   14. natfriend: Extensions for making SIP a NAT-friendly protocol.
SIPPING WG Charter
   1. PSTN and/or 3G telephony-equivalent applications that need a standardized approach (PSTN)
        informational guide to common call flows
        support for T.38 fax
        requirements from 3GPP for SIP usage
        framework of SIP for telephony (SIP-T)
        call transfer and call forwarding
        AAA application in SIP telephony
        mapping between SIP and ISUP


   2. Messaging-like applications of SIP (Msg)
        support for hearing-/speech-impaired calling
        development of usage guidelines for subscribe-notify (RFC 2848, SIP events) to ensure commonality among applications using them,
         including SIMPLE WG's instant messaging.
   3. Multi-party applications of SIP (MP)
        the working group will review a number of technical pieces including call transfer, subscribe-notify, SIP features negotiation, and session
         description protocol (SDP) capability negotiation, and will develop requirements and an initial design or framework for multi-party
         conferencing with SIP.
   4. SIP calling to media servers (Media)
        the working group will develop a requirements draft for an approach to SIP interaction with media servers. An example is whether a
         voicemail server is just a box that a caller can send an INVITE to.
        At a later time, the working group and chairs may request of the Area Directors that new tasks be added to the charter. Such additions to the
         charter will require IESG approval.
        The group will work very closely with SIP working group. The group will also maintain open dialogue with the IPTEL working group, whose
         Call Processing Language (CPL) related to the task areas in a number of ways. The group will also coordinate closely with SIMPLE, AAA,
         and MMUSIC (SDP development)
• WIRELESS + PRESENCE
• XML




      4
Contact Centers
 Phone Customer
                                        inbound:                    Contact Center Systems
                                        8YY, 900,
                                        Local

                                     •TTS
                                     •ASR
                                                           VoiceXML VRU
    Agents                           •DTMF            telephone #                        URL
                                                                                            Forms/Data
                                      VoiceXML pages
                 web
                content                                                                           Backend
                                                                 Web site
                                                                                                   Data
                                                          Pages/CGI/Servlets
                                                                                                   Bases
  Web Customer


             Generald M. Karam, “General Issues Surrounding VoiceXML Products,” SIP Summit 2001 Fall, 2001. 9.
    Application Server Component Architecture
                Application   AAA and                                                          Credit Card
                                                                     Location
                Service        Policy              Dialing
                                                                                               Verification
                Providers                           Plans             Service
                               Server

                                                               Service          Presence      Transaction
                      IVR          Web                        Controller         Server         Server


                                             SIP             SIP                                         SIP Clients
  SIP Clients                                Srv             Srv


                           IP Network
Callers              SIP and HTTP Messages            All real time IP communication services are             Called
                                                       implemented by SIP and HTTP call flows
                           RTP Media


     Gateways                                                                                           Gateways

   PSTN                                                                                                        PBX

                    Text to      Directory    UM Store         Conference       Media Mixer    Streaming
                    Speech       & ENUM       Voice Mail        Scheduler                       Content

                                                                                      출처: VoN
VoIP addressable market may top $6
billion by 2006

Up from $180        180
million in 2001     160                                        Middle East/Africa
                    140                                        Asia/Pacific
Majority of
                    120
revenues will be    100
                                                               Central Asia
                                                               C. & E. Europe
international and    80                                        Western Europe
new minutes         60                                         Latin America/
                                                               Caribbean
                    40
Demand for VoIP                                                North America
                    20
services is truly
globalizing          2000   2001   2002   2003   2004   2005 2006

                            PC-to-phone users (Millions)

 Source: Ovum
                                                        출처: VoN
전망




     Ref: VoIP pros and cons, by Edwin Mier, Network World, 08/27/01
      http://www.nwfusion.com/research/2001/0827featside5.html
The Growing Number of SIP Clients
(1,000,000s)




                          출처: VoN
Nuclear Winter - Drive

Evolution > Revolution (?)
Killer application => late in 2003
2 leaders emerge



                     COCKROACH
Who is Supporting/Using SIP

 Microsoft               Hardware Vendors
                            MIP Telecom
 AOL/Time Warner
                            Sonus
 Carriers                  Cisco
  PointOne                 Convedia
  Level 3                  PingTel
  WorldCom               Software Vendors
 CASP (Communications      Pactolus
  ASP)                      SS8
  Vonage                   NexTone
                            Dymanicsoft
  GoBeam
                            Sylantro
Technology Partners
 PROXIES      APPLICATION SERVERS   PRESENCE SERVERS   SIP PHONES




     SOFTSWITCHES




           GATEWAYS




                                               출처: VoN
THANK YOU




        Thank you

				
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