2 - SIP y H.323: Establecimiento y gestión de sesiones multimedia Asterisk

Transcripción

1 2 - Voz sobre IP (VoIP) SIP y H.323: Establecimiento y gestión de sesiones multimedia Asterisk Computer Networking: A Top Down Approach Featuring the Internet, 3 rd edition. Jim Kurose, Keith Ross Addison-Wesley, July Thanks to : RADCOM technologies H. Shulzrinne Paul. E. Jones (from packetizer.com) TECNOLOGÍAS DE RED Master IC

2 Voice-over-Data (VoD) Enables New Applications Click to talk web sites for e-commerce Digital white-board conferences Broadcast audio and video over the Internet or a corporate Intranet Integrated messaging: check (or leave) voice mail over the Internet Instant messaging Voic notifications Stock notifications Callback notification Fax over IP Etc. 2

3 3 Sesion Initiation Protocol SIP is end-to-end, client-server session signaling protocol SIP s primarily provides presence and mobility Protocol primitives: Session setup, termination, changes,... Arbitrary services built on top of SIP, e.g.: Redirect calls from unknown callers to secretary Reply with a webpage if unavailable Send a JPEG on invitation Features: Textual encoding (telnet, tcpdump compatible). Programmability. Post-dial delay: 1.5 RTT Uses either UDP or TCP Multicast/Unicast comm. support

4 SDP codecs Application RTSP SIP RTP Transport TCP UDP Where s SIP DNS(SRV) Network IP Physical/Data Link Ethernet 4

5 Are true Internet hosts Choice of application Choice of server IP appliances Implementations 3Com (3) Columbia University MCI WorldCom (2) Mediatrix (1) 4 Nortel (4) Siemens (5) 2 IP SIP Phones and Adaptors 1 Analog phone adaptor Palm control

6 6 User Agents SIP Components UAC (user agent client): Caller application that initiates and sends SIP requests. UAS (user agent server): Receives and responds to SIP requests on behalf of clients; accepts, redirects or refuses calls. Server types Redirect Server Accepts SIP requests, maps the address into zero or more new addresses and returns those addresses to the client. Does not initiate SIP requests or accept calls. Proxy oyservere Contacts one or more clients or next-hop servers and passes the call requests further. Contains UAC and UAS. Registrar Server A registrar is a server that accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles. Location Server Gateways Provides information about a caller's possible locations to redirect and proxy servers. May be co-located with a SIP server. A Sip Gateway service allows you to call 'real' numbers from your software or have a dedicated 'real' telephone number which comes in via VoIP

7 DNS Server Outgoing Proxy SIP DNS SIP SIP Trapezoid Location Server Registrar Incoming Proxy SIP SIP Originating User Agent SIP RTP Terminating User Agent 7

8 DNS Server DNS SIP SIP Triangle? Location Server Registrar Incoming Proxy SIP SIP Originating User Agent SIP RTP Terminating User Agent 8

9 SIP Peer to Peer! SIP Originating Terminating User Agent RTP User Agent TECNOLOGÍAS DE RED 9

10 INVITE ACK OPTIONS CANCEL SIP Methods Requests a session Final response to the INVITE Ask for server capabilities Cancels a pending request BYE Terminates a session 1 0 REGISTER Sends user ss address to server

11 SIP Responses 1XX Provisional 100 Trying 2XX Successful 200 OK 3XX Redirection 302 Moved Temporarily 4XX Client Error 404 Not Found 5XX Server Error 504 Server Time-out 6XX Global Failure 603 Decline 1 1

12 User A Calls Talking INVITE: sip: ACK RTP User B Ringing Rings OK Answers Talking SIP Flows - Basic Hangs up BYE OK 1 2

13 INVITE sip:e9-airport.mit.edu SIP/2.0 From: "Dennis 1c41 To: sip:e9-airport.mit.edu Call-Id: Cseq: 1 INVITE Contact: "Dennis Baron"<sip: @ > Content-Type: application/sdp Content-Length: 304 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE SIP INVITE Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/ (WinNT) Date: Thu, 30 Sep :28:42 GMT Via: SIP/2.0/UDP

14 IETF RFC 2327 Session Description Protocol SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. SDP includes: The type of media (video, audio, etc.) The transport protocol (RTP/UDP/IP, H.320, etc.) The format of the media (H.261 video, MPEG video, etc.) Information to receive those media (addresses, ports, formats and so on) 1 4

15 v=0 o=pingtel 5 5 IN IP s=phone-call c=in IP t=0 0 m=audio 8766 RTP/AVP a=rtpmap:96 eg711u/8000/1 a=rtpmap:97 eg711a/8000/1 a=rtpmap:0 pcmu/8000/1 SDP a=rtpmap:8 pcma/8000/1 a=rtpmap:18 g729/8000/1 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000/1 1 5

16 GIPS Enhanced G.711 8kHz sampling rate Voice Activity Detection Variable bit rate G.711 8kHz sampling rate 64kbps G.729 8kHz sampling rate 8kbps Voice Activity Detection CODECs 1 6

17 User B REGISTER: sip:dbaron@mit.edu REGISTER: (add credentials) Unauthorized OK Registrar MIT.EDU Location MIT.EDU SIP Flows - Registration sip:dbaron@mit.edu Contact

18 REGISTER sip:mit.edu SIP/2.0 From: "Dennis To: "Dennis Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 5 REGISTER SIP REGISTER Contact: "Dennis Baron"<sip: @ ;LINEID=05523f7a97b54dfa3f0c0e3746d73a24> Expires: 3600 Date: Thu, 30 Sep :46:53 GMT Accept-Language: en Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/ (WinNT) Content-Length: 0 Via: SIP/2.0/UDP

19 SIP/ Unauthorized From: "Dennis To: "Dennis Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 5 REGISTER Via: SIP/2.0/UDP SIP REGISTER 401 Response Www-Authenticate: Digest realm="mit.edu", nonce="f b8ae841b9b0ae8a92dcf0b ", opaque="reg:change4" Date: Thu, 30 Sep :46:56 GMT Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, NOTIFY, SUBSCRIBE, INFO User-Agent: Pingtel/2.2.0 (Linux) Accept-Language: en Supported: sip-cc-01, timer Content-Length: 0 1 9

20 REGISTER sip:mit.edu SIP/2.0 From: "Dennis To: "Dennis Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 6 REGISTER SIP REGISTER with Credentials Contact: "Dennis Baron"<sip: @ ;LINEID=05523f7a97b54dfa3f0c0e3746d73a24> Expires: 3600 Date: Thu, 30 Sep :46:53 GMT Accept-Language: en Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/ (WinNT) Content-Length: 0 Authorization: DIGEST USERNAME=" @mit.edu", REALM="mit.edu", NONCE="f b8ae841b9b0ae8a92dcf0b ", URI="sip:mit.edu", RESPONSE="ae064221a50668eaad1ff2741fa8df7d", OPAQUE="reg:change4" Via: SIP/2.0/UDP

21 User A Calls INVITE: sip:dbaron@mit.edu ACK Trying Ringing OK Proxy MIT.EDU INVITE: sip:dbaron@ Ringing User B Rings OK Answers SIP Flows Via Proxy Talking RTP Talking Hangs up BYE OK 2 1

22 User A Calls Talking INVITE: sip:joe@mit.edu ACK Proxy MIT.EDU INVITE: sip:38400@ Gateway SIP Flows Via Gateway Trying Rings Ringing OK RTP ACK Ringing OK Answers Talking Hangs up BYE BYE OK OK 2 2

23 2 3 SIP INVITE with Record-Route INVITE sip:37669@ SIP/2.0 Record-Route: <sip: :5080;lr;a;t=2c41;s=b07e28aa8f94660e a44b9ed50> From: \"Dennis Baron\"<sip: @mit.edu>;tag=2c41 To: sip:37669@mit.edu Call-Id: call @ Cseq: 1 INVITE Contact: \"Dennis Baron\"<sip: @ > Content-Type: application/sdp Content-Length: 304 Accept-Language: en Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/ (WinNT) Date: Thu, 30 Sep :44:30 GMT Via: SIP/2.0/UDP :5080;branch=z9hG4bK2cf12c563cec06fd1849ff799d069cc0 Via: SIP/2.0/UDP ;branch=z9hG4bKd26e44dfdc d9d32a143a7f4d8 Via: SIP/2.0/UDP Max-Forwards: 17

24 Just a sampling of IETF standards work IETF RFCshttp://ietf.org/rfc.html SIP Standards RFC3261 Core SIP specification obsoletes RFC2543 RFC2327 SDP Session Description Protocol RFC1889 RTP - Real-time Transport Protocol RFC2326 RTSP - Real-Time Streaming Protocol RFC3262 SIP PRACK method reliability for 1XX messages RFC3263 Locating SIP servers SRV and NAPTR RFC3264 Offer/answer model for SDP use with SIP 2 4

25 2 5 SIP Standards (cont.) RFC3265 SIP event notification SUBSCRIBE and NOTIFY RFC3266 IPv6 support in SDP RFC3311 SIP UPDATE method eg. changing g media RFC3325 Asserted identity in trusted networks RFC3361 Locating outbound SIP proxy with DHCP RFC3428 SIP extensions for Instant Messaging RFC SIP REFER method eg. call transfer SIMPLE IM/Presence - g p SIP authenticated identity management -

26 NATs: Hole Punching - Peers tras distinto NAT TECNOLOGÍAS DE RED 2 6

27 Terminals Multipoint Control Units (MCUs) Gateways Gatekeeper Border Elements Elements of an H.323 System Referred to as endpoints 2 7

28 Telephones Video phones IVR devices Voic Systems Soft phones (e.g., NetMeeting ) Terminals 2 8

29 MCUs Responsible for managing multipoint conferences (two or more endpoints engaged in a conference) The MCU contains a Multipoint Controller (MC) that manages the call signaling and may optionally have Multipoint Processors (MPs) to handle media mixing, switching, or other media processing 2 9

30 Gateways The Gateway is composed of a Media Gateway Controller (MGC) and a Media Gateway (MG), which may co-exist or exist separately The MGC handles call signaling and other non-mediarelated functions The MG handles the media Gateways interface H.323 to other networks, including the PSTN, H.320 systems, and other H.323 networks (proxy) 3 0

31 Gatekeeper The Gatekeeper is an optional component in the H.323 system which is primarily used for admission control and address resolution The gatekeeper may allow calls to be placed directly between endpoints or it may route the call signaling through itself to perform functions such as follow- me/find-me and forward on busy 3 1

32 Border Elements and Peer Elements Peer Elements, which are often co-located with a Gatekeeper, exchange addressing information and participate in call authorization within and between administrative domains Peer Elements may aggregate address information to reduce the volume of routing information passed through the network Border Elements are a special type of Peer Element that exists between two administrative domains Border Elements may assist in call authorization/authentication directly between two administrative domains or via a clearinghouse 3 2

33 The Protocols (cont) H.323 is a framework document that describes how the various pieces fit together H defines the call signaling between endpoints and the Gatekeeper RTP/RTCP (RFC 3550) is used to transmit media such as audio and video over IP networks H Annex G and H.501 define the procedures and protocol for communication within and between Peer Elements H.245 is the protocol used to control establishment and closure of media channels within the context of a call and to perform conference control 3 3

34 The Protocols (cont) H.450.x is a series of supplementary service protocols H.460.x is a series of version-independent independent extensions to the base H.323 protocol T.120 specifies how to do data conferencing T.38 defines how to relay fax signals V defines how to relay modem signals H.235 defines security within H.323 systems X.680 defines es the ASN.1 syntax used by the Recommendations X.691 defines the Packed Encoding Rules (PER) used to encode messages for transmission on the network 3 4

35 Registration, Admission, and Status - RAS Defined in H Allows an endpoint to request authorization to place or accept a call Allows a Gatekeeper to control access to and from devices under its control Allows a Gatekeeper to communicate the address of other endpoints Allows two Gatekeepers to easily exchange addressing information i 3 5

36 3 6 Registration, Admission, and Status RAS (cont) T RRQ RCF (endpoint is registered) ARQ ACF (endpoint may place call) DRQ DCF GK Symbol Key: (call has terminated) T Terminal GK GW Gatekeeper Gateway

37 The H323 stack TECNOLOGÍAS DE RED 3 7

38 O.S. Client Price Windows NetMeeting ti +/- free Unix (Linux) DC-Share nv Sun Sunforum +/- free H323 Clients You can find a bigger list at: 3 8

39 2- Voz sobre IP (VoIP) SIP y H.323: Establecimiento y gestión de sesiones multimedia Asterisk TECNOLOGÍAS DE RED Master IC

40 ASTERISK Aplicación de software libre que implementa los servicios de una centralita telefónica de VoIP. Permite conectar teléfonos de VoIP (que también pueden ser programas de ordenador o softphones ), fax, líneas RDSI, líneas telefónicas analógicas convencionales Inicialmente i desarrollada d para Linux pero actualmente t existen versiones para casi todas las plataformas. trixbox (con t minúscula) es una distribución Linux (en concreto de CentOS) que incluye Asterisk y FreePBX que es un entorno gráfico basado en WEB para una configuración cómoda y más sencilla de Asterisk. 4 0

41 Soporta SIP, H.323, MGCP, IAX Se obtiene de : ftp://ftp.digium.com Integra casi todos los codecs de audio Soporte de Telefonía Tradicional Soporte de Telefonía por Voz IP ASTERISK APIs para desarrollo de nuevos servicios y aplicaciones Integración con Bases de Datos Integración con Aplicaciones ya desarrolladas Código Abierto: sw libre 4 1

42 IAX (Inter-Asterisk exchange) Actualmente en la versión 2 (IAX2) es un protocolo que aborda el problema de los NATs. Utilizar el mismo puerto UDP para la señalización y para la transmisión de los datos (RTP). Simplifica el número de agujeros (hole-punching) a realizar en el NAT para que el interlocutor en la intranet sea alcanzable desde d Internet. t Algunos autores abogan porque IAX será el futuro de VoIP y otros plantean que la regulación en tema de NATs, o incluso su desaparición con la entrada de IPv6 dejaran a SIP en su posición de liderato. 4 2

43 Configuración básica TECNOLOGÍAS DE RED 4 3

44 Configuración básica (2) TECNOLOGÍAS DE RED 4 4

45 Configuración básica (3) TECNOLOGÍAS DE RED 4 5

46 IMPLEMENTACIÓN DE TELEFONÍA IP EN UNA ORGANIZACIÓN INTEGRACIÓN CISCO-ASTERISK TECNOLOGÍAS DE RED 4 6

47 CARACTERISTICAS CISCO CALL MANAGER Solución de Telefonía IP de Cisco Distribuible Escalable (30000 lineas/servidor) id Soporta muchos usuarios Sobre Windows o linux Soporta gran variedad de teléfonos 4 7

48 Sip H323 MGCP (Megaco Protocol) PROTOCOLOS TECNOLOGÍAS DE RED 4 8

49 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV 0 * OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7960 messages services i directories settings 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV 0 * OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7960 messages services i directories settings 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV 0 * OPER 3 DEF 6 MNO 9 WXYZ # 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV 0 * OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7960 messages services i directories settings CISCO IP PHONE 7960 messages services i directories settings 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV 0 * OPER 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV 0 * OPER 3 DEF 6 MNO 9 WXYZ # 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7960 CISCO IP PHONE 7960 messages services messages services i i directories settings directories settings TECNOLOG GÍAS DE RED OBJETIVO FINAL 4 9

50 FUNCIONAMIENTO DE CALL MANAGER TECNOLOGÍAS DE RED 5 0

51 CONFIGURACIÓN CM Interfaz Web TECNOLOGÍAS DE RED 5 1

52 PARTITIONS Dividen el conjunto de route patterns en subconjuntos de destinos alcanzables identificados por un nombre. Una partición contiene una lista de Route Patterns Facilitan el enrutado de llamadas dividiendo el route plan en subconjuntos lógicos que se pueden basar en la organización, localización y tipo de llamada 5 2

53 Partitions TECNOLOGÍAS DE RED 5 3

54 SEARCH SPACES Es una lista ordenada de rutas de partición. Estas rutas se asocian a los dispositivos (teléfonos). Determinan las particiones que los dispositivos que hacen una llamada buscan para que esta llamada se realice 5 4

55 ROUTE PATTERNS String de digitos y un conjunto de acciones La llamada al destino se hace solo si se marca la secuencia correcta definida en el route pattern Se pueden usan caracteres especiales (x ) para hacer rangos, etc Definir route patterns para diferentes tipos de llamadas: nacionales, sin salida. 5 5

56 ESQUEMA DE NUMERACIÓN 67xxx: Teléfonos IP HW (Vera) 68xxx: SoftPhones 69xxx: Teléfonos SIP 7xxxx: Teléfonos analógicos (fuera del Call Manager) 11xxx: Teléfonos móviles 5 6

57 Route patterns TECNOLOGÍAS DE RED 5 7

58 Debe haber uno por cada campus Otro que será el router de salida general. Coste: euros GATEWAYS 5 8

59 Gateways TECNOLOGÍAS DE RED 5 9

60 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV * 0 OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7960 messages services i directories settings 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV * 0 OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7960 messages services i directories settings 1 2 ABC 4 5 GHI JKL 7 8 PQRS * TUV 0 OPER 3 DEF 6 MNO 9 WXYZ # 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV * 0 OPER 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE messages services i directories settings CISCO IP PHONE 7960 messages services i directories settings 1 2 ABC 4 5 GHI JKL 7 8 PQRS TUV * 0 OPER 2 ABC 4 5 GHI JKL 7 8 PQRS * TUV 0 OPER 3 DEF 6 MNO 9 WXYZ # 3 DEF 6 MNO 9 WXYZ # CISCO IP PHONE 7960 CISCO IP PHONE 7960 messages services messages services i i directories settings directories settings TECNOLOG GÍAS DE RED Es un enlace desde el Call Manager al Asterisk: se enrutan llamadas de uno al otro Se define mediante la IP del Asterisk TRUNK CON ASTERISK 6 0 1

61 Trunk TECNOLOGÍAS DE RED 6 1

62 TELEFONOS un identificador, el Device Name (3 caracteres más la dirección MAC ) una descripción (ej. la persona asociada) el pool al que corresponde su estado (registrado o no) la dirección IP del teléfono: sólo se muestra si el teléfono está registrado 6 2

63 Teléfonos TECNOLOGÍAS DE RED 6 3

64 Teléfonos II TECNOLOGÍAS DE RED 6 4

65 Teléfonos III Teléfono Cisco Teléfono SIP 300 Euros Euros Configuración desde el CM SIP_ADDITIONAL.CONF 6 5

66 [69001] < Extensión username=69001 < Podría ser el login type=friend record_out=adhoc record_in=adhoc dh qualify=no port=5060 nat=never < Su buzón de voz asociado (en el voic .conf) host=dynamic dtmfmode=info context=from-internal canreinvite=no callerid=device <69001> language=es Teléfonos IV 6 6

67 Softphone Cisco IP Communicator Teléfonos V TECNOLOGÍAS DE RED 6 7