SIP: Protocolo para los servicios multimedia del futuro

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1 SIP: Protocolo para los servicios multimedia del futuro Joaquín Salvachúa

2 Requisitos de aplicaciones VR HD videoconf VideoConf VoD HD-VoD BW NFS/SMB MM-IM Web DBMS Text-IM Calendar Latencia - 2 -

3 Visión General H.323 MGCP/Megaco SDP SIP Signaling Quality of Service RTSP reservation RSVP Media transport Media encaps (H.261,MPEG) RTP Physical link network transport PPP Sonet TCP IPv4,IPv6 AAL3/4 AAL5 ATM UDP Ethernet PPP V.34

4 SIP, H.323 y MGCP Control de la llamada H.323 Señalización Flujos multimedia Media Audio/ Video H.245 Q.931 H.225 RAS SIP MGCP MEGACO RTP RTCP RTSP TCP UDP IP - 4 -

5 Arquitectura Terminal: Dispositivo final del usuario. Gateway: Traducctor entre mundos no H.323 Gatekeeper: Gestion de una zona, identificación y recursos. MCU: Control y flujos multiusuario

6 Protocolos Gatekeeper Annex G Q.931/H.245 Gatekeeper RAS Q.931/ H.245 Q.931/ H.245 RAS Endpoint Signalling (Q.931) H.245 RTP/RTCP Endpoint Gatekeeper Routed Signaling Direct Routed Signaling - 6 -

7 Aplicaciones Text telephony H.324 Multimedia H.320 Multimedia H.323 Multimedia T.120 Data conferencing T.140 T.140 Voice and video T.140 Voice and video T.140 Voice and video T.140 Compatibility equalizers Transparent AL1 H.245 H.224 Client 2 TCP H.245 T.134 T.124 GCC V.18 H.223 V.34/V.80 H.221 Network access H Network access PSTN PSTN ISDN IP Network T.123 Any Network Gateway functions, with transparent transmission of T.140 data between the different T.140 data channel types. T

8 Terminal Video I/O Equipment Video Codec H.261, H.263 Audio I/O Equipment User Data Applications T.120, etc Audio Codec G.711, G.722 G.723, G.728 G.729 Receive Path Delay (jitter buffer) H Layer Local Area Network Interface System Control H.245 Control System Control User Interface Call Control H RAS Control H

9 SIP: Session Initiation Protocol Estándar del IETF (RFC 3261) Protocolo genérico de establecimiento de sesiones multimedia. Interoperabilidad con anteriores VoIP. Diseñado específicamente para IP e Internet: similar a HTTP. Permitirá la aparición de nuevos servicios y aplicaciones Escalable y flexible: 4Bajo coste de establecimiento de llamada.

10 Protocolos de tiempo real Los protocolos definidos por el IETF 4Control de flujos y sesión: 4Descripción de los flujos 4Transporte de datos t. Real: 4Reserva de recursos (QoS): SIP, RTSP SDP RTP/RTCP RSVP,DiffServ

11 Grupos relevantes del IETF Audio/Video Transport (avt) - RTP Differentiated Services (diffserv) QoS in backbone IP Telephony (iptel) CPL, GW location, TRIP Integrated Services (intserv) end-to-end QoS Media GatewayControl (megaco) IP telephony gateways Multiparty Multimedia Session Control (mmusic) SIP, SDP PSTN and Internet Internetworking (pint) mixt services Resource Reservation Setup Protocol (rsvp) Service in the PSTN/IN Requesting InTernet Service (spirits) Session Initiation Protocol (sip) signaling for call setup Signaling Transport (sigtran) PSTN signaling over IP Telephone Number Mapping (enum) surprises! Instant Messaging and Presence Protocol (impp)

12 Evolución de Estandares Tiphon TIPIA ANSI T1 avt ITU-T IMTC ETSI IETF SG16 mmusic SG15 ATM Forum STQ pint SG11 SG2 MSF sigtran VOP TC Sec megaco inow! aaa NA2 iptel PacketCable TINA

13 La dirección SIP es similar a la de correo. No tiene porque ser IP: Por ejemplo URL son: sip:

14 Componentes SIP Agentes de Usuario: 4 Cliente que inicial las llamadas 4 Modo cliente y servidor. Servidores 4 Proxy: Statefull: Con estado Stateless: Sin estado. 4 Redirect 4 Registration Registration Server Redirect Server Proxy Server Proxy Server UAC UAS

15 Es de Igual a Igual: Peer 2 Peer SIP Components User Agent User Agent

16 Un sistema típico SIP Components Location Server Redirect Server Registrar Server PSTN User Agent Proxy Server Proxy Server Gateway

17 A un Sistema completo V Gateway V Gateway Phone Marshal Server (Gateway) Provisioning Server Redirect Server(s) Feature Server(s) CDR Server(s) Policy Server(s) Marshal Server Internet COPS/OSP RADIUS SNMP Network Manager H.323 Terminal Marshal H.323/SIP Server Translator Marshal Server SIP IP Phones 3 rd Party Billing System Marshal Server MGCP/SIP Translator Clearing House MGCP Device

18 Telephone Telephone switch T1/E1 RTP/SIP Cisco 2600 gateway sipc Despliegue SIP sipconf SIP conference server sipd SIP proxy, redirect server rtspd RTSP media server sipum SIP/RTSP Unified messaging SQL database RTSP e*phone Quicktime RTSP clients Web server Hardware Internet (SIP) phones Web based configuration sip323 NetMeeting Software SIP user agents SIPH.323 convertor H.323

19 Similar a HTTP

20 Mensajes SIP Todos los componentes usan mensajes SIP: SIP Requests: 4INVITE Initiates a call by inviting user to participate in session. 4ACK - Confirms that the client has received a final response to an INVITE request. 4BYE - Indicates termination of the call. 4CANCEL - Cancels a pending request. 4REGISTER Registers the user agent. 4OPTIONS Used to query the capabilities of a server. 4INFO Used to carry out-of-bound information, such as DTMF digits SIP Responses: 41xx - Informational Messages. 42xx - Successful Responses. 43xx - Redirection Responses. 44xx - Request Failure Responses. 45xx - Server Failure Responses. 46xx - Global Failures Responses.

21 Establecimiento de una comunicación SIP 1.Registering, initiating and locating the user. 2.Determine the media to use involves delivering a description of the session that the user is invited to. 3.Determine the willingness of the called party to communicate the called party must send a response message to indicate willingness to communicate accept or reject. 4.Call setup. 5.Call modification or handling example, call transfer (optional). 6.Call termination

22 Realización de una llamada lts.ncsc.mil DNS Location server telcordia.com Proxy INVITE INVITE Ringing 200 OK ACK Proxy INVITE Ringing 200 OK Linda Media Streams Peter INVITE SDP proposes media type(s), IP & ports to send to 200 OK SDP accepts/rejects media, gives IP & ports to send to

23 Detalladamente User A Proxy 1 Proxy 2 User B INVITE F1 407 Proxy Authorization F2 ACK F3 INVITE F4 (100 Trying) F6 180 Ringing F OK F14 ACK F15 BYE F OK F21 INVITE F5 (100 Trying) F8 180 Ringing F OK F13 ACK F16 Both way RTP BYE F OK F22 INVITE F7 180 Ringing F9 200 OK F12 ACK F17 BYE F OK F23 Reference: A. Johnston, S. Donovan, R. Sparks, C. Cunningham, K. Summers: SIP Telephony Call Flow Examples. Work in progress, IETF, October

24 Servicios más complejos Service Features: Call Hold Consultation Hold Unattended Transfer Call Fwd Unconional Call Fwd on Busy Call Fwd on No Answer 3-Way Conference Single Line Extension Find-Me Incoming Call Screening Outgoing Call Screening Secondary Number In Secondary Number Out Do Not Disturb Call Waiting

25 Reenvio SIP User A SIP Proxy 1 SIP User B1 SIP User B2 Invite F1 (100 Trying) F3 Invite F2 486 Busy Here F4 Ack F5 Invite F6 180 Ringing F8 200 OK F Ringing F7 200 OK F9 ACK F11 ACK F12 Both Way RTP Established (A-B2) Bye F13 Bye F OK F OK F

26 Ejemplo 2 Bob INVITE INVITE phone1.office.com Alice REGISTER INVITE vm.office.com

27 Ejemplo 1 Bob phone1.office.com; CANCEL Alice 200 OK 200 OK RTP/RTCP v-mail vm.office.com; SETUP rtspd

28 Ejemplo 3 Bob phone1.office.com Alice BYE 200 OK RTP to v-mail vm.office.com

29 Ejemplo 4 Bob phone1.office.com Alice INVITE RTP Quick-time v-mail vm.office.com RTSP SETUP/PLAY rtspd

30 Soporte para movilidad Mobile Host 4 5 SIP Proxy Server Foreign Network SIP Redirect Server Home Network 1 INVITE moved temporarily 3, 4 INVITE Corresponding Host 3 6 5, 6 OK 7 Data

31 SIMPLE: Extensiones de presencia Nueva entidad: Presence Agent 4 Purely logical entity 4 Knows presence state of user 4 Receives SUBSCRIBE requests 4 Generates NOTIFY requests 4 Co-located with proxy/registrar or User Agent Operaciones 4 Subscriber send SUBSCRIBE 4 Routed to PA using normal SIP 4 PA authorizes subscriber 4 Acceptance contains presence state 4 NOTIFY sent when state changes Routed using SIP Record-Route SUBSCRIBE Subscriber NOTIFY Proxy REGISTER Presence Agent + Proxy/Registrar = Presence Server Presentity

32 SIMPLE (SIP for IM and Presence) lts.ncsc.mil telcordia.com Presence server Proxy SUBSCRIBE SUBSCRIBE NOTIFY NOTIFY Proxy Update Presence Linda Peter Linda subscribes to notifications of changes in Peter s status: Off-line, on-line, busy, away, available,...

33 Extension de IM Operation of Extension 4 Messages carried in SIP messages 4 New method - MESSAGE 4 Routed to recipient using normal SIP techniques 4 Simple extension Can also use SIP to establish messaging session 4 On same level as RTP 4 Can apply SIP tools multiparty, transfers, etc. Features 4 Associates an IM with an existing call 4 Any MIME data can be sent 4 Routed by existing proxies and registrars 4 Possible to have a different client for IM and communications

34 Auto-Conference Controller initiates SUBSCRIBE session with presence server Media Server Conference Server On NOTIFYs update presence When all online Send IM to some with HTTP URL for accept Use 3pcc to connect others to MS MS fetches VoiceXML from AS HTTP POST of each accept/reject Use 3pcc to connect each to same conference URL Application Server Presence Server Presence HTTP IM SIP Call

35 Multiconferencia User 1 User 2 User 3 IM Conf. Srvr User 1 INVITEs User 2 4 User 2 accepts 4 They IM User 1 decides to add user 3 User 1 connects himself to IM conference server User 1 reconnects IM stream with user 2 to conference server User 1 calls user 3 and connects their IM stream to conference server INV 200 ACK INV 200 ACK INV w/o SDP 200 INV 200 ACK w/ SDP ACK INV w/o SDP 200 INV 200 ACK ACK w/ SDP

36 Integración Integración de VoIP con: 4Web 4 4Presencia y mensajería instantánea 4Chats textuales 4Juegos Interactivos voz Juegos IM Web Comunicaciones interactivas Presencia Chat video Telefonía por Internet

37 Necesidad Azar y necesidad: 4Aparecen redes de banda ancha por aplicaciones 4Aparecen nuevas aplicaciones que utilizan redes de banda ancha. Concepto de killer application superado: 4Killer Cocktail No hay aplicaciones que sustituyen totalmente a todas 4Teoria del TAMBIEN

38 MGCP

39 Components Call agent or media gateway controller 4Provides call signaling, control and processing intelligence to the gateway. 4Sends and receives commands to/from the gateway. Gateway 4Provides translations between circuit switched networks and packet switched networks. 4Sends notification to the call agent about endpoint events. 4Execute commands from the call agents. Call Agent or Media Gateway Controller (MGC) MGCP SIP H.323 Call Agent or Media Gateway Controller (MGC) MGCP Media Gateway (MG) Media Gateway (MG)

40 What media gateways do Connection control 4Unicast 4Multicast 4Circuit to packet (IP) 4Circuit to packet (ATM) 4Packet to packet 4Circuit to circuit Loopback testing The ability to identify/request endpoint attributes 4The media protocol used (RTP, fax-protocol,...) 4The payload type (e.g. codec), 4The codec-related attributes like packetisation interval, jitter buffer size and silencesuppression whereappropriate 4The generation of comfort noise during silent periods

41 What media gateways do(2) The ability to identify/request endpoint attributes 4The application of encryption/decryption and identification of the encryption schemes. 4Theecho cancellation 4The lawful interception of the content of a specified media stream Content insertion 4Playing tone or announcement (IVR) 4Mute request 4Continuity testing, etc., as required by SS7 and others Event detection 4On/off hook 4DTMF Association management

42 Simplified Call Flow When Phone A goes off-hook Gateway A sends a signal to the call agent. Gateway A generates dial tone and collects the dialed digits. The digits are forwarded to the call agent. The call agent determines how to route the call. The call agent sends commands to Gateway B. Gateway B rings phone B. The call agent sends commands to both gateways to establish RTP/RTCP sessions. Call Agent Media Gateway Controller MGCP Gateway A RTP/RTCP MGCP Gateway B Analog Phone A Analog Phone B

43 MGCP Commands Call Agent Commands: 4EndpointConfiguration 4NotificationRequest 4CreateConnection 4ModifyConnection 4DeleteConnection 4AuEndpoint 4AuConnection Gateway Commands: Gateway Commands: 4Notify 4DeleteConnection 4RestartInProgress

44 Characteristics of MGCP MGCP: MGCP: 4A master/slave protocol. Assumes limited intelligence at the edge (endpoints) and intelligence at the core (call agent). Used between call agents and media gateways. Differs from SIP and H.323 which are peer-to-peer protocols. 4Interoperates with SIP and H

45 MGCP, SIP and H.323 MGCP divides call setup/control and media establishment functions. MGCP does not replace SIP or H.323. SIP and H.323 provide symmetrical or peer-to-peer call setup/control. MGCP interoperates with H.323 and SIP. For example, 4 A call agent accepts SIP or H.323 call setup requests. 4 The call agent uses MGCP to control the media gateway. 4 The media gateway establishes media sessions with other H.323 or SIP endpoints. In this example, an H.323 gateway is decomposed into: A call agent that provides signaling. A gateway that handles media. MGCP protocol is used to control the gateway. H.323 Gateway Call Agent/ Media Gateway Controller MGCP H.323 H.323 Gateway Media Gateway Media RTP/RTCP

46 Example Comparison H A user picks up analog phone and dials a number. 2. The gateway determines how to route the call. 3. The two gateways exchange capabilities information. 4. The terminating gateway rings the phone. 5. The two gateways establish RTP/RTCP session with each other. MGCP 1. A user picks up analog phone and dials a number. 2. The gateway notifies call agent of the phone (endpoint) event. 3. The Call agent determines capabilities, routing information, and issues a command to the gateways to establish RTP/RTCP session with other end. 1 Analog Phone H.323 Gateway RTP/ RTCP H.323 Gateway 4 Analog Phone 1 Analog Phone 2 Gateway A Call Agent/ Media Gateway Controller RTP/ RTCP Gateway B Analog Phone

47 What is Megaco? A protocol that is evolving from MGCP and developed jointly by ITU and IETF: 4Megaco - IETF. 4H.248 or H.GCP - ITU. For more information, refer to: 4IETF - 4Packetizer

48 Gateway control protocol evolution, roughly SGCP (Bellcore) MGCP (Bellcore) H.GCP/megaco/etc. (ITU-T, IETF) SDCP (Level3 TAC) MDCP (Lucent)

49 Comparación H.323 SIP MEGACO H.248 Nuevos Servicios No Si No Respuesta Lenta OK OK Escalable No Si No Integración con Internet No Si No

50 SoftSwitch App Server Communications SIP Jain / Parlay LDAP Interoperability Source: Softswitch Protocols & Actg Server Interoperability Prov Server Network Centric User Centric Application Centric Application Servers Customized Call Processing & Treatment H.323 <-> SIP Mediation Module H.323 Gatekeeper Module Dir Map Server Interoperability IP Client-SoftSwitch Communications SIP (eg native, or MS XP) Authentication User Profiling Presence Mgmt Personalized Call Treatment C4 C5 SIP Proxy Server Module Wiretap Feature Servers SIP < -> H.248 Mediation Module IP Centrex IP Call Centres Conferencing Mutli-lingual IVR MegaCo Session Mgr Module FoIP Msg Store Unified. Communications Service Creation Engine Mobility LNP Mgmt 800 SS7 / CCS7 Signaling Gateway SDN Svcx Module HLR/VLR MGCP Controller Module Unified Messaging Network Mgmt OAM&P OSS Integration Activation Interoperability Media Gateways Inter-SoftSwitch Communications SIP-T BICC Interoperability Softswitch Media Gateway -SoftSwitch Communications SIP H.323 H.248 / MegaCo MGCP Media Gateway SDN SS7 Network 800 LNP

51 NAT Traversal Network Address Translators (NATs) map a private IP address space to externally visible (public) IP addresses 4Conserve scarce public IP addresses 4Shield internal hosts from outside world Useful for enterprises, cable modem networks, broadband access routers, internet cafes NATs interfere with peer-to-peer protocols such as SIP 4SIP clients must identify the IP address and ports they will use to receive media streams (in payload of their signaling messages) 4But they don t know their externally visible addresses One of the SIP community s biggest problems

52 STUN: NAT traversal STUN Server Private Network A Internet Private Network B NAT NAT STUN Client SIP Client STUN Client SIP Client SIP Proxy/Registrar Source: P. Thermos, Telcordia

53 Network Evolution Process Proprietary 1st Gen IP Proprietary 2nd Gen IP Open, Distributed Systems P R O P R I E T A R Y Services & Applications Call Control & Switching Transport Hardware Services Gatekeeper H.323 Gateway Services Semi- Softswitch (Hardswitch) Media Gateway Services, Applications & Features (Management, Provisioning and Back Office) Open Protocols APIs Softswitch Call Control Open Protocols APIs Transport Hardware Solutions come from a single vendor that supplied everything in one proprietary box: software, hardware and applications Customers were lockedin to their vendor Solutions can come from multiple vendors, at all levels who supply open standards-based products Open standards enable innovation and reduce costs

54 Circuit-switch vs. Softswitch Circuit-Switched Soft-Switched P R O P R I E T A R Y Services & Applications Call Control & Switching Transport Hardware Services, Applications & Features (Management, Provisioning and Back Office) Open Protocols APIs Softswitch Call Control Open Protocols APIs Transport Hardware Solutions come from a single vendor that supplied everything in one proprietary box: software, hardware and applications Customers were locked-in to their vendor no room for innovation, expensive to implement and maintain Solutions can come from multiple vendors, at all levels who supply open standards-based products Customers are free to choose best-in-class products to build their network. Open standards enable innovation and reduce costs

55 Lo interesante son los servicios Services econstruction Service Schedule Finance Au Video Video Finance XML/SOAP econstruction Portal Schedule Network XML/SOAP P2P, Open Souce Au Network User A User B Web Services User B Third Wave User A

56 Valor de la comunicación SIP SIP mobility, location manager, caller preferences... La integración de todos los servicios es el killer-cocktail commercial quality voice QoS Referencia: Voz Jonathan Rosenberg/DynamicSoft Telephony Voz Telephony and voice mail web Integrated voice and data communications voice mail web presence New services voice mail web presence QoS Commercial quality voice mail web presence QoS security e-commerce

57 SIP/RTP Media Architecture CAS, Q.931, SS7 MGCP Telephony Gateway is a SIP client PCM SG MG Dumb Network Public IP Backbone Goes everywhere End-to-end control Consistent for all services DNS mobility Messaging SIP Web Directory RTP Security QoS Media services Sessions Telephony Any other sessions

58 Servidores de aplicaciones

59 Call Processing Language (CPL) : Designed by the IETF to support sophisticated telephony services 4 May be used by both SIP or H.323. XML based scripting language for describing controlling call services [3] 4 Simple Syntax 4 Extendible 4 Easily eed by GUI tools Scripts runs on network SIP signaling server to create end user services 4 Lightweight CPL interpreter is need to parser & validate scripts

60 CPL Example : A simple script that blocks anonymous callers [4] ; <?xml version="1.0"?> <!DOCTYPE cpl PUBLIC "-//IETF//DTD RFCxxxx CPL 1.0//EN" "cpl.dtd"> <cpl> <incoming> <address-switch field="origin" subfield="user"> <address is="anonymous"> <reject status="reject" reason="i don't accept anonymous calls" /> </address> </address-switch> </incoming> </cpl>

61 Common Gateway Interface (CGI) : Almost identical to HTTP CGI [5] Language independent ( Perl, Tcl, C, C++,... ) 4 Any binary may be executed as a separate program Suitable for services that contains substantial web content Passes message parameters through environmental variables to a separate program. 4 More flexible but more risky Feb. 1, 2001: RFC 3050 (Common Gateway Interface for SIP) published [6]

62 Java Servlets : Similar to HTTP servlets Instead of using a separate process, messages are passed to a class The class runs within a JVM (Java Virtual Machine) on server Security provided by Java Portable between OSs & servers

63 JAIN SIP API: Low level API that maps to IETF - RFC 2543 Interfaces for services across circuit switched and packet networks Three major objectives : 4 Service Portability Write Once Run Anywhere 4 Network Converges Any underlying network architecture IP, ATM,Wireless,... 4 Service Provider Access by Anyone

64 JAIN SIP API: ( Cont.) Three SIP APIs under JAIN initiative; 4 JAIN SIP API (JSR 32) : Low level API for almost any signaling protocol ( SIP, H.323,... ) Requires extensive knowledge of SIP. Avaliable at Final Release, 4 JAIN SIP Lite : High Level API for rapid application development Especially User Agent development Under development, 4 SIP Servlets : [7] API for SIP servlets Under development,

65 Untrusted third party applications JAIN Parlay APIs JAIN Service Provider Access (JSPA) Secure network region The JAIN APIs JAIN Service Creation Environment (JSCE) JAIN Service Logic Execution Environment (JSLEE) APIs Trusted third party applications Security interface Jain Call Control (JCC) and JAIN Coordination and Transaction (JCAT) TCAP ISUP INAP MAP SIP H.323 MGCP Megaco

66 JAIN SPA Applications written in Java, Java Beans and Enterprise Java Beans can leverage the write once, run anywhere paradigm Customer & Application Profiles JAIN SPA APIs Open Network Applications OSA/Parlay Client IP Network Secure Telco Boundary OSA/Parlay Server Customer and Service Profiles JCC Application Service Plane MM Apps JAIN SLEE API Customer and Service Profiles Java Call Control API Feature/App Server ISUP API ISUP Stack Softswitch MGCP API MGCP Stack SIP API SIP Stack Control Plane

67 Conclusiones SIP permite una transición. Transición de la VoIP actual a la futura. Unificación de dos mundos: 4Internet: Flexible y ágil 4Telefonía tradicional: Sólido y robusto Toma de lo mejor de ambos mundos. Protocolo Clave para la Internet-NG

68 Opinión No habrá negocio en vender llamadas más baratas. El negocio estará en servicios aún por inventar: necesidad de prototipado rápido. Cada usuario necesitará sus propio conjunto de servicios que interoperen. Toolkits de servicios son el futuro y sus lenguajes. H.323 /MGCP será sustituido por SIP. ICQ sustituido por SIMPLE. SHIP opensource:

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