Multimedia over Internet  Paper 1 H. Schulzrinne, "A comprehensive multimedia control architecture for the Internet", Proc. of the Int. Workshop on Network and Operating System Support for Digital Audio and Video (NOSSDAV), (St. Louis, Missouri), May1997.  Paper 2 W. Jiang, J. Lennox, H. Schulzrinne and K. Singh, "Towards Junking the PBX: Deploying IP Telephony", in Proc. International Workshop on Network and Operating System Support for Digital Audio and Video (NOSSDAV), (Port Jefferson, New York), Jun. 2001

Outline  Overview of Internet  Paper 1 Introduction Session Initiation Protocol (SIP) Real-Time Stream Protocol (RTSP) Combining SIP and RTSP Description of Multimedia Presentations  Paper 2 Telephone Network IP telephony Architecture PSTN Inter-operability Other Issues  Summary and Future Work

Overview of Internet OSITCP/IP Application Presentation Session Transport NetworkInternet Data Link Subnet Physical

Overview of Internet  The physical layer This layer defines the type of physical signals ( electrical, optical, etc.), as well as the type of media (wires, coaxial cable, satellite, etc.).  The data link layer Common examples of data link control protocols are the HDLC, SDLC, and PPP.  The network layer ( Internet Protocol – IP)  The transport layer ( TCP/UDP)  The application layer Telnet, ftp, SMTP, HTTP, NNTP, LDAP, Several multimedia protocols ( SIP, RTP, H.323, etc. )

OSPF Overview of Internet PingFTPH.323SIPRTSPRSVP S/MGCP /NCS RTP/ RTCP Telnet TCPUDP IPIGMPICMPARPRARP Link Layer

Paper 1 A comprehensive multimedia control architecture for the Internet Henning Schelzrinne Dept. of Computer Science Columbia University New York, NY 10027

Introduction  In this paper, he present two independent, but interacting protocols that initiate and control stored, live and interactive multimedia sessions in the Internet.  The protocols support the following scenarios: Phone call Invitation to a multi-party conference Near video-on-demand Video-on-demand Virtual presentations Distributed digital editing Combining stored, live and interactive multimedia

Introduction  Session Initiation Protocol (SIP) Inviting participants to a multimedia session Establish and control multimedia conferences  Real-Time Stream Protocol (RTSP) Control playback and recording for stored continuous media Control delivery of stored and live streaming multimedia content

Session Initiation Protocol (SIP)  Conference control applications use SIP to invite humans and media servers into a multicast conference or establish a two-party phone call.  The conference initiation phase has to accomplish three goals: locate the terminal (phone, workstation, mobile phone, answering machine, … ) where the called party can be reached. agree on a set of media and possible encodings for communication determine if the called party wants to be reached.

Major Features of SIP  User location Determination of the end system to be used for communications.  User capabilities Determination of the media and media parameters to be used.  User availability Determination of the willingness of the called party to engage in communications.  Call setup Establishment of call parameters at both called and calling party.  Call handling Including transfer and termination of calls.

Names and Addresses  A name is an identification of an entity ( independent of its physical location), such as a person, and applications program, or even a computer.  An address is also an identification but it reveals additional information about the entity, principally information about its physical or logical placement in a network.  The IP Address ( 32 bits ) Network (7 bits) Local address ( 24 bits) Network address (14 bits) Local (8 bits) Local address (16 bits) Network address (21 bits) 1110Multicast address (28 bits) Class A Class B Class C Multicast format

Session Initiation Protocol (SIP)  SIP chose an -like identifier of the form or  The domain name can be either the name of the host that a user is logged in at the time, an address or the name of a domain- specific translation service.

SIP address resolution Address is SIP server? forward? accept ? Address is SMTP server? Busy, no answer reject success failure Get address (VRFY,EXPN) same as before ? Send MIME message Y N Y N Y N N Y N Y

SIP Redirect Server CALL bob play.cse.psu.edu 302 moved temporarily Location: INVITE play alice cse.psu.edu Redirect Server Location Server 200 OK

SIP Proxy Server CALL play alice cse.psu.edu Proxy Server bob Location Server play.cse.psu.edu INVITE 200 OK

SIP Forking Proxy CALL run alice cse.psu.edu Proxy Server bob Location Server run.cse.psu.edu jump.cse.psu.edu INVITE 200 OK jump INVITE

Other issues  Choosing Terminals Many people have several ways of being reached, including a telephone, , fax, or a pager. A SIP server can return a descriptive list of alternative terminals, their capabilities and addresses.  Locating Callees In a local area, a person may move around from terminal to terminal. A SIP can work over a connectionless transport protocol and multicast a “search” for a particular party.  Negotiating Media Types and Encodings The SIP INVITE request to join a conference or phone call contains a listing of the media types and associated encodings that the calling party is willing to use. The called party simply responds with a subset of media types and encodings that it is willing to use.

Real-Time Stream Protocol (RTSP)  RTSP initiates and controls delivery of stored and live multimedia content to both unicast and multicast destinations.  The Real-Time Protocol (RTP) is designed to support real time traffic, which provides services that include payload type identification, sequence numbering, time-stamping, and delivery monitoring.  The Real-Time Control Protocol (RTCP) is a control component. Both data sender and receivers periodically multicast RTCP messages to monitor network quality.

Real-Time Protocol (RTP) Transit Network Transit Network 512 kbit/s translator 384 kbit/s

Real-Time Protocol (RTP) Transit Network Transit Network 64 kbit/s each mixer 64 kbit/s ( Combine 192 kbit/s to 64 kbit/s )

The basic operation of RTSP client web server media server HTTP GET session description SETUP PLAY PAUSE TEARDOWN RTP audio RTP video RTCP

The basic operation of RTSP  First, the client should obtain a description of the multimedia presentation. The description can be retrieved by HTTP or ftp. It can be different format.  Then the client will initiates a session with the SETUP request to the media server. The SETUP request also indicates where the server is to send the data, if not provided in the presentation description.  The presentation itself can be controlled with PLAY, RECORD and PAUSE.  The client closed the session with the TEARDOWN request.

Description of Multimedia Presentations  A data structure to describe the session or presentation they are initiating and controlling.  SDF describes presentations as a hierarchy of sequential, alternative and time-parallel streams.  The design of SDF found its way into a proposed SGML-based description called RTSL.  RTSL is intentionally purely descriptive and contains no scripting functionality.

Sample RTSP session description Twister <track type=audio e=“PCMU/8000/1” src=“rtsp://audio.example.com/twister/audio.en/lofi”> <track type=audio e=“DVI4/16000/2” pt=“90 DVI4/8000/1” src=“rtsp://audio.example.com/twister/audio.en/hifi”> <track type=“video/jpeg” src=“rtspu://video.example.com/twister/video”>

Combining SIP and RTSP Internet conferencing example Combining SIP and RTSP

Possible client conferencing architecture using SIP and RTSP

Paper 2 Towards Junking the PBX: Deploying IP Telephony Wenyu Jiang, Jonathan Lennox, Henning Schelzrinne and Kundan Singh {wenyu,lennox, Dept. of Computer Science Columbia University New York, NY 10027

Telephon e Network  The goals of a telephone system There had to be sufficient direct current flow to operate the customer’s station sets. Support dc/low-frequency call process signaling (dialing, ringing) and to keep the signaling simple at the customer’s terminal. Limit signal loss to acceptable levels such that the voice conversation between the customers would appear as “natural” as possible.  The telephone dialing plan NXX-XXXX N: 2~9 X: 0~9

Example of a call Local Station Originating Office Local Station Terminating Office Idle Dial tone Idle Line connect Dial pulsing Trunk connect Start dial Dial pulsing Ringing Ringback Answer Busy

IP Telephony  Internet telephony is defined as the transport of telephone calls over the Internet.  Internet telephony integrates a variety of services provided by the current Internet and the Public Switched Telephone Network (PSTN) infrastructure.  Internet telephony employs a variety of protocols, including RTP, H.323, MGCP, Megaco, SIP, etc.

IP Telephony  Services and prices vary, with these offerings: PC-to-PC calls PC-to-telephone calls Telephone-to-telephone calls Fax service Voice messaging  Examples of Service Providers: Deltathree ( ) Net2Phone ( MediaRing ( Dialpad ( PhoneFree (

Architecture

Architecture  SIP server SIP proxy, redirect and registration server  SQL database storing the current network addresses and phone numbers where the user can be reached.  PSTN gateway connect the PBX to the LAN with a T1 trunk  User agents allow users to interact with the system over IP  Media server storage and delivery of announcements and voice mail messages  Unified messaging centralized answering machine and voice mail system  Conference server centralized audio/video conference server  SIP-H.323 translator a signaling gateway between SIP and H.323

User Database  Every user of the system is given a unique identifier of the form also called a canonical user  The user information is stored in the SQL database as the Primary User Table and indexed by the user identifier.  There are other tables in the MySQL database: contact table alias table

Incoming Calls

 Transform the callee address to a canonical user identifier for database look up host portion: erlang.cs.columbia.edu—cs.culumbia.edu User name portion: alias  name mapping  dial plan  Retrieves user, contact, and policy information Proxied or redirected Authentication Forking proxy

PSTN Inter-Operation  Dialplan On the gateway, define a voice over IP call-leg specifier(called a dial peer) Direct-inward-dialing (DID) mode No-DID mode  Connecting to the PBX Modeusageadvantages DIDDial directlySimpler dialing from PSTN No-DIDDial extensionSupports more users

PSTN Inter-Operation  Security Issues User registrations user registrations need to be authenticated to prevent unauthorized users from redirecting calls to themselves or elsewhere. Remote callers a local user may choose to force remote callers to be authenticated. Our authentication goal is to establish a consistent mapping between a caller’s SIP identity and her identity. This ensures that the SIP caller is indeed identical to the corresponding address. Access to the PSTN we need to restrict access to the PSTN gateway to prevent “free” calls.

Other Services  Programmable Call Handling The XML based Call Processing Language (CPL) The SIP Common Gateway Interface  Unified Messaging Centralized voice mail RTSP for storage and retrieval of voice messages  Multi-Party Conferencing A SIP conference server with audio and video capabilities The canonical user identifier The dynamic modification of the dialplan The SQL database stores various conference attributes

Scalability  Multiple conference servers can be installed, with each running only tens of active conferences.  For scaling proxy servers, make use of the DNS SRV capability in SIP.

Summary  Paper 1 describe two protocols that support the multimedia conference, namely the SIP to establish and control multimedia conferences and the RTSP to control delivery of stored and live streaming multimedia content.  Paper 2 describes the architecture of the Internet telephony installation: SIP server SIP-PSTN gateway RTSP media server unified messaging server conferencing server SIP-H.323 translator.

Future Work  The Internet multimedia architecture is still missing two pieces, namely a floor control protocol and a shared drawing protocol.  “embedded application” that work behind the scenes of web pages, games and virtual reality.  Continue with integration of additional services.  Build highly scalable systems.  A commercial deployment involves many other issues related to security, billing and quality of service.