Multimedia over Internet

Paper 1H. 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 2W. 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

OutlineOutline

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 InternetOverview of Internet

OSIOSI TCP/IPTCP/IPApplication

ApplicationPresentation

Session

Transport Transport

Network Internet

Data LinkSubnet

Physical

Overview of InternetOverview of Internet

The physical layerThis 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 layerCommon 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 layerTelnet, ftp, SMTP, HTTP, NNTP, LDAP, Several multimedia protocols ( SIP, RTP, H.323, etc. )

OSPF

Overview of InternetOverview of Internet

Ping FTP H.323 SIP RTSP RSVPS/MGCP

/NCSRTP/RTCP

Telnet

TCP UDP

IP IGMPICMPARP RARP

Link Layer

Paper 1Paper 1

A comprehensive multimedia control architecture for the Internet

Henning Schelzrinne

Schulzrinne@cs.columbia.edu

+1 212 939 7042

Dept. of Computer Science

Columbia University

New York, NY 10027

IntroductionIntroduction

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 callInvitation to a multi-party conferenceNear video-on-demandVideo-on-demandVirtual presentationsDistributed digital editingCombining stored, live and interactive multimedia

IntroductionIntroduction

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 Session Initiation Protocol (SIP)(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 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 AddressesNames 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)0

10

110

Local address ( 24 bits)

Network address (14 bits)

Local (8 bits)

Local address (16 bits)

Network address (21 bits)

1110 Multicast address (28 bits)

Class A

Class B

Class C

Multicast format

Session Initiation Protocol Session Initiation Protocol (SIP)(SIP)

SIP chose an email-like identifier of the form

user@domain or user@IP_address. The domain name can be either the name of

the host that a user is logged in at the time, an email address or the name of a domain-specific translation service.

SIP address resolutionSIP address resolution

Address is SIP server?

forward?

accept ?

Address is SMTP server?

Busy, no answerreject

success

failure

Get address(VRFY,EXPN)

same as before ?

Send MIME message

Y

N

Y

N

Y

N

N

Y

N

Y

SIP Redirect ServerSIP Redirect Server

CALL bob@cse.psu.edu

bob

play.cse.psu.edu

302 moved temporarily

Location: bob@play.cse.psu.edu

INVITE bob@play.cse.psu.edu

play

alice

bob@play

cse.psu.edu

Redirect Server

Location Server

200 OK

SIP Proxy ServerSIP Proxy Server

CALL bob@cse.psu.edu

play

alice

bob@play

cse.psu.edu

Proxy Server

bob

Location Server

play.cse.psu.edu

INVITE bob@play.cse.psu.edu

200 OK

200 OK

SIP Forking ProxySIP Forking Proxy

CALL bob@cse.psu.edu

run

alice

bob@jump

cse.psu.edu

Proxy Serverbob

Location Server

run.cse.psu.edujump.cse.psu.edu

INVITE bob@run200 O

K200 OK

jump

INV

ITE bob@jum

p

Other issuesOther issues

Choosing TerminalsMany people have several ways of being reached, including a telephone, email, fax, or a pager. A SIP server can return a descriptive list of alternative terminals, their capabilities and addresses.

Locating CalleesIn 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 EncodingsThe 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 Real-Time Stream Protocol (RTSP)(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)Real-Time Protocol (RTP)

TransitNetwork

TransitNetwork

512 kbit/s

translator

384 kbit/s 384 kbit/s

Real-Time Protocol (RTP)Real-Time Protocol (RTP)

TransitNetwork

TransitNetwork

64 kbit/s each

mixer

64 kbit/s 64 kbit/s

( Combine 192 kbit/s to 64 kbit/s )

The basic operation of RTSP

client

webserver

mediaserver

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

<title>Twister</title><session> <group language=en lipsync> <switch> <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”>

</switch> <track type=“video/jpeg”

src=“rtspu://video.example.com/twister/video”> </group></session>

Combining SIP and RTSP

Internet conferencing example Combining SIP and RTSP

Combining SIP and RTSP

Possible client conferencing architecture using SIP and RTSP

Paper 2Paper 2

Towards Junking the PBX: Deploying IP Telephony

Wenyu Jiang, Jonathan Lennox, Henning Schelzrinne and Kundan Singh

{wenyu,lennox, hgs,kns10}@cs.columbia.edu

Dept. of Computer Science

Columbia University

New York, NY 10027

Telephon e NetworkTelephon 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-XXXXN: 2~9

X: 0~9

Example of a callExample of a call

Local StationOriginating

OfficeLocal Station

TerminatingOffice

Idle

Dial tone

IdleIdle

Line connect

Dial pulsing Trunk connect

Start dial

Dial pulsingRinging

Ringback

AnswerAnswer

Answer

Busy

IP TelephonyIP 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 TelephonyIP Telephony

Services and prices vary, with these offerings: PC-to-PC calls PC-to-telephone calls Telephone-to-telephone calls Fax service E-mail Voice messaging

Examples of Service Providers: Deltathree (www.deltathree.com ) Net2Phone (www.Net2Phone.com) MediaRing (www.mediaring.com) Dialpad (www.dialpad.com) PhoneFree (www.phonefree.com)

ArchitectureArchitecture

ArchitectureArchitecture

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 DatabaseUser Database

Every user of the system is given a unique identifier of the form user@domain, also called a canonical user identifier.

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 CallsIncoming Calls

Incoming CallsIncoming 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: aliasname mappingdial plan

Retrieves user, contact, and policy information Proxied or redirected Authentication Forking proxy

PSTN Inter-OperationPSTN 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

ModeMode usageusage advantagesadvantages

DIDDID Dial directly Simpler dialing from PSTN

No-DIDNo-DID Dial extension Supports more users

PSTN Inter-OperationPSTN Inter-Operation

Security Issues User registrations

user registrations need to be authenticated to prevent unauthorized users from redirecting calls to themselves or elsewhere.

Remote callersa 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 email identity. This ensures that the SIP caller is indeed identical to the corresponding email address.

Access to the PSTNwe need to restrict access to the PSTN gateway to prevent “free” calls.

Other ServicesOther 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

ScalabilityScalability

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.

SummarySummary

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 WorkFuture 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.