Multimedia over Internet

1. 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

2. 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

3. Overview of Internet

4. 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. )

5. Overview of Internet Ping Telnet FTP H.323 SIP RTSP RSVP S/MGCP /NCS RTP/ RTCP TCP UDP OSPF ARP ICMP IP IGMP RARP Link Layer

6. Paper 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

7. 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

8. 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

9. 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.

10. 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.

11. 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 ) Class A 0 Network (7 bits) Local address ( 24 bits) 10 Network address (14 bits) Local address (16 bits) Class B 110 Network address (21 bits) Local (8 bits) Class C Multicast format 1110 Multicast address (28 bits)

12. Session Initiation Protocol (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.

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

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

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

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

17. Other issues • Choosing Terminals Many 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 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.

18. 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.

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

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

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

22. 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.

23. 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.

24. 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>

25. Combining SIP and RTSP Internet conferencing example Combining SIP and RTSP

26. Combining SIP and RTSP Possible client conferencing architecture using SIP and RTSP

27. Paper 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

28. 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

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

30. 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.

31. IP 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)

32. Architecture

33. 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

34. User 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

35. Incoming Calls

36. 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

37. 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

38. 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 email identity. This ensures that the SIP caller is indeed identical to the corresponding email address. • Access to the PSTN we need to restrict access to the PSTN gateway to prevent “free” calls.

39. 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

40. 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.

41. 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.

42. 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.