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Columbia InterNet Extensible Multimedia Architecture

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Columbia InterNet Extensible Multimedia Architecture Powered By Docstoc
					                                                                                                                  CINEMA (Columbia InterNet Extensible Multimedia Architecture)
                                                                      presented by – Kundan Singh, Joint work with Wenyu Jiang, Jonathan Lennox, Sankaran Narayanan, Henning Schulzrinne, Xiaotao Wu
                                                                                                         More information at http://www.cs.columbia.edu/IRT/cinema/

                                Approach                                                                                                                                                           Project Objectives                                                                                                                                                 Performance
             Develop protocols (SIP, RTSP, RTP,…)                                                                                                                                                                                                                                                                          sipstone: benchmark for SIP servers
                                                                                                                                             A flexible architecture to support clients and servers for wide                                                                                                                Different signaling vs. media components
             Implement common reusable libraries
             Provide distributed servers components                                                                                         range of multimedia communication applications such as video                                                                                                                   Black-box measurement and white-box profiling
                                                                                                                                                                                                                                                                                                                            Load balancing, thread pooling, and reactive
             Integrate with web, email, phone systems                                                                                       conferencing, Internet telephony/radio, interactive voice                                                                                                                       system to improve performance
                                                                                                                                             response, unified messaging, presence and multimedia                                                                                                                           Novel peer-to-peer IP telephony using SIP
               Session Initiation Protocol (SIP)-based                                                                                       collaboration.
                   enterprise VoIP infrastructure
                                                                                                                                                                                                                                                                                                                                            Load sharing and failover in SIP
                                                                                                                                                                                                                                                                                                                              example.com                                                        Second stage proxy/registrar (sipd)
                                                                                  CINEMA servers                                                                                                                                                                                       P                 P                    _sip._udp
                                                                                                                                                                                                                                                                                                                                SRV 0 0 s1
     Local/long distance                   Telephone                       sipconf:             rtspd: media server                                          Unified messaging using SIP                                                           P2P VoIP                                                                     SRV 0 0 s2                   First stage stateless
                                                                                                                                                                                                                                                                                                                                                             proxy server farm
                                                                                                                                                                                                                                                                                                                                                                                                 a1                  M         a.example.com
     e.g., 1-212-5551212                                                   conference server                                RTSP                                                                                                                                                       P                 P                      SRV 0 0 s3                                                                                     _sip._udp
                                           switch
                                                                                                                                                             and RTSP                                                                              using SIP                                                                    SRV 1 0 ex                                                                                      SRV 0 0 a1
                                                                                                                                   RTSP clients                                                             2                                                                                    P                                                              s1                               a2                             SRV 1 0 a2
  PSTN                                      Department                                                                                                                                             sipd
                                                                                                                                                                                                                                                                                                                                                                                                                     S
                                                                                                                                   e.g., Quicktime                                     1                                Bob’s phone
                                            PBX
                                                                                                                sipum:                                                                                            5                                Peer-to-peer Internet telephony avoids the
                                                                                                                                                                                           6                                                                                                                                  sip:bob@example.com
                                                                                                                unified                                                                                         2            sipum                 configuration and maintenance cost of                                                                                     sip:bob@b.example.com
        Internal                                               sipd:                                                                                                                                                                                                                                                                                            s2
        Telephone                                              proxy,
                                                                                                                messaging                                              Alice’s phone
                                                                                                                                                                                               7                                                   server-based architecture and dependency
        e.g., 7040          713x                               redirect,                                                                                                                                        4                                  on controlled infrastructure such as DNS. We
                                                               registrar
                                                                                                   SQL           cgi
                                                                                                                         Web                                                                                                      3                use Chord algorithm on top of SIP for an                                                                     s3                               b1                  M         b.example.com
                                                                                                                                                             1. Alice (caller) calls Bob                                                                                                                                                                                                                                       _sip._udp
                                                                                                                                                                                                                                                   interoperable, scalable and robust P2P-SIP
                                                                                                 database               server
                                                                                                                                                             2. The SIP server forks the call to Bob’s phone                                                                                                                                                                                                                    SRV 0 0 b1
                          SIP/PSTN Gateway
                            e.g., Cisco 2600
                                                                                                                 vxml                   Web based
                                                                                                                                                                and the mail server                                                                endpoint.                                                                                                                                     b2                  S          SRV 1 0 b2
                                                                                                                                       configuration
                                                                                                                                                             3. After 10 seconds, the mail server sets up                     rtspd
                                                                                                     SIP                                                        RTSP sessions to playback welcome
                                                                                                                                                                message and to record mail
                                                                                                         VXML
                                                                                                                                                             4. Mail server accepts the call
                                                                                                                                                                                                                                                                                                                                                            Web                                             Web
             7134
                                                                                                                                                             5. SIP server cancels the other branch                                              Presence and event notification                                                                           scripts                                         scripts
                                                                                                                                                             6. SIP server forwards the acceptance                                                                            office.com
                                   7136                                                                                      H.323
                                                                                                                                                                                                                                                                                                             bob@office.com                                                D1                                                      D2
                                                                                   siph323:                                                                  7. Media packets are sent directly between the
                                                                                                                                                                                                                                                                               Presence server
                                                                                   SIP-H.323                                                                    RTSP server and caller                                                                                                                                                                                    Master                 Bi-directional                 Slave
              alice@cs.columbia.edu                                                                                                                                                                                                                                                                                     PUA
                                                                                   translator                                                                                                                                                                                       PA                                                                                    Slave                    replication                  Master
                    (software phone)                                                                                                 H.323 clients                                                                                          alice@home.com                                           REGISTER
                                                                                                                                     e.g., NetMeeting                                                                                                                                                                                                      P1                                              P2
                                                                                                                                                                                                                                                               SUBSCRIBE

                                                                                                                                                                                                                                                                   NOTIFY
                                     Multimedia conferencing                                                                                                                                                                                                                     registrar
                                                                                                                                                                                                                                                                                                                     PUA
                                                                                                                                                                                                                gatekeeper
                                                                                                                                              SIP-H.323                                sipd                                                                                                                                                             phone.cs.columbia.edu                           sip2.cs.columbia.edu
                                                       A SIP/RTP-based centralized conference server                                                                                                                                                                                                                PUA + PA                                                                      REGISTER
                          SIP323       Low
                                                       to support audio mixing, video forwarding, text                                        gateway
  High
                                       bitrate
                                                       chat and screen sharing among heterogeneous
                                                                                                                                                                                   SIP                      H.323                                                                                                                                      _sip._udp
                                                                                                                                                                                                                                                                                                                                                          SRV 0 0 5060 phone.cs.columbia.edu                                    proxy1 = phone.cs

                                                       endpoints such as PC and phones. It has play-
                                                                                                                                                                                                                                                                                                                                                          SRV 1 0 5060 sip2.cs.columbia.edu                                     backup = sip2.cs
  quality
                                                                                                                                              A signaling translator between ITU-T’s multistage H.323
                                                       out delay adjustment for wide area Internet,                                           and IETF’s SIP that supports different dialing modes, has
                                                       web-based conference setup, high quality
                                                       audio (G.722, G.711) as well as low bit rate                                           a built-in gatekeeper and is transparent to media path.                                                                   Overview
                                                       codecs (GSM, DVI).                                                                                                                                                                        Multimedia communication
                                                                                                                                                                                                                                                                                                                                   Multimedia application components
                                                                                                                                                                                                                                                                                                                                                                                                                              Interactive
        SIP/PSTN
                                                                                                                                                                                                                                                    Audio, video, text, screen sharing, …                                                                  Internet               Internet
                                                                                                                                                                                                                                                                                                                                                                                                                            voice response
                                                                                                                                                                    Programmable SIP proxy                                                          PSTN interworking, IVR                                                                                Telephony              Radio/TV
                                                                                                                Programmable IP                                                                                                                  Multi-devices
                                                                                                                                                                                                                                                                                                                                                                                                          Messaging
                                                                                                                                                                                                                                                                                                                                                                                                         and Presence
                                           Interactive voice                                                    telephony services                                                                                                                  IP-phone, telephone, X10, Ncast, …                                                          Video
                                                                                                                                                                                                                                                                                                                                                                                                                                            Unified
                                                                                                                                                                                                                                                                                                                                                                                                                                           messaging
PSTN phone
                  SIP/PSTN gateway          response (IVR)                                                      Programmable call routing based                                                                                                  Collaboration
                                                                                                                                                                                                                                                                                                                                             conferencing

                                                                                                   vxml         on time of day, caller id, etc., using                                     cgi
                                                                                                                                                                                           CPL                                                      Voicemail, discussion forum,…                                                                                                                                                       Media

                            Call request                                                   Web server
                                                                                                                server side Call processing                                                         SQL                                                                                                                                                                                                                                 G.711
                                                                                                                                                                                                                                                                                                                                                                                                                                        MPEG

                                                                                         CGI, servlet, JSP      language, Common Gateway                                                                                                                                                                                                                                   SIP
                                                                                                                                                                                                                                                                                                                                                                                      RTSP
                                                                                                                                                                                                                                                                                                                                                                                                  SAP       RSVP         RTCP
                                                                                                                                                                                                                                                                                                                                     Application layer
      SIP phone
                                                                                                                interface (CPL), Java servlets or                                                                                                                                                                                                                                                                                       RTP
                                                                                                                                                                                                                                                               Other Applications
                                                 SIP-based VoiceXML                                             client side Language for End                                                                                                                                                                                      Transport (TCP, UDP)

                                                  browser (sipvxml)
                                                                                                                System services (LESS) scripts                               Libraries (C/C++)                                                                                                                                    Network (IPv4, IPv6)                               Signaling            Quality of service       Media transport
                                           Press 1 to listen to next message,             Media server
              SIP phone                                                                                                                                                                                                           RTSP server                                                                                           Link layer
                                           2 to forward …
                                                                                                                                                                        SIP, RTP, audio mixing, DB
                                                                                                                                                                        interface, SNMP interface,                                      RTSP API              SIPUA          SIP
                                                                                                                                                                                                                                                                                                                                      Physical layer
                                                                                                                                                                        RTSP, DNS SRV/NAPTR,                                                                    API         proxy
                                                                                                                                                                                                                                                                                                                                                                         Program
                                                                                                                                                                                                                                                                                                                                                                                      Voice                              Speech/
                                                                                                                                                                                                                                                                                                                                                                            Call                  DTMF      Mixing                       SDP
                                                                                                                                   IP endpoint                          win32 portability,…                             RTP              RTSP tr
+1 212 9397040
                                       PSTN interworking                                                                                                                                                              Interface
                                                                                                                                                                                                                                                 SIP transaction
                                                                                                                                                                                                                                                                                                                                                                          routing
                                                                                                                                                                                                                                                                                                                                                                                      XML                                  text


                                                                                                                                                        sip:wenyu@cs                                                                                                   Client Branch

                              Telephone
                                                                                                                                                                                                                                         HTTP Message Parsing
                                                                                                                                                                                                                                                                                                                               … moving from IP telephony to
   Telephone                                               SIP/PSTN                      SIP server
                                                                                                                                                                                                                                           Transport layer (TCP/UDP)
   subscriber                 network
                                                           gateway                         (sipd)                                                                                                                                                                                                                             real-time multimedia collaboration…
                                                                                                                                     sip:7141@cs.columbia.edu
                                                                                                                                                                                                          Layered Architecture

				
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