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2 - Voz sobre IP ( VoIP )

Computer Networking: A Top Down Approach Featuring the Internet , 3 rd edition. Jim Kurose, Keith Ross Addison-Wesley, July 2004. . 2 - Voz sobre IP ( VoIP ) . SIP y H.323: Establecimiento y gestión de sesiones multimedia Asterisk . Thanks to : RADCOM technologies H. Shulzrinne

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2 - Voz sobre IP ( VoIP )

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  1. Computer Networking: A Top Down Approach Featuring the Internet, 3rd edition. Jim Kurose, Keith RossAddison-Wesley, July 2004. 2 -Voz sobre IP (VoIP) SIP y H.323: Establecimiento y gestión de sesiones multimedia Asterisk • Thanks to : • RADCOM technologies • H. Shulzrinne • Paul. E. Jones (from packetizer.com)

  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 • Voicemail notifications • Stock notifications • Callback notification • Fax over IP • Etc.

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

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

  6. SIP Components • User Agents • 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 Server • 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 • Provides information about a caller's possible locations to redirect and proxy servers. May be co-located with a SIP server. • Gateways • 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. SIP Trapezoid Location Server DNS Server DNS Registrar SIP Outgoing Proxy Incoming Proxy SIP SIP SIP SIP Originating User Agent Terminating User Agent RTP

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

  9. SIP Peer to Peer! SIP Originating User Agent Terminating User Agent RTP

  10. SIP Methods • INVITE Requests a session • ACK Final response to the INVITE • OPTIONS Ask for server capabilities • CANCEL Cancels a pending request • BYE Terminates a session • REGISTER Sends user’s 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

  12. User A User B “Calls” 18.18.2.4 ACK 200 - OK INVITE: sip:18.18.2.4 180 - Ringing 200 - OK BYE Rings Answers RTP Talking Talking Hangs up SIP Flows - Basic

  13. SIP INVITE INVITE sip:e9-airport.mit.edu SIP/2.0 From: "Dennis Baron"<sip:6172531000@mit.edu>;tag=1c41 To: sip:e9-airport.mit.edu Call-Id: call-1096504121-2@18.10.0.79 Cseq: 1 INVITE Contact: "Dennis Baron"<sip:6172531000@18.10.0.79> 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/2.1.11 (WinNT) Date: Thu, 30 Sep 2004 00:28:42 GMT Via: SIP/2.0/UDP 18.10.0.79

  14. Session Description Protocol • IETF RFC 2327 • “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)

  15. SDP v=0 o=Pingtel 5 5 IN IP4 18.10.0.79 s=phone-call c=IN IP4 18.10.0.79 t=0 0 m=audio 8766 RTP/AVP 96 97 0 8 18 98 a=rtpmap:96 eg711u/8000/1 a=rtpmap:97 eg711a/8000/1 a=rtpmap:0 pcmu/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:18 g729/8000/1 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000/1

  16. CODECs • 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

  17. Registrar Location User B MIT.EDU MIT.EDU 200 - OK 401 - Unauthorized REGISTER: (add credentials) REGISTER: sip:dbaron@MIT.EDU sip:dbaron@MIT.EDU Contact 18.18.2.4 SIP Flows - Registration

  18. SIP REGISTER REGISTER sip:mit.edu SIP/2.0 From: "Dennis Baron"<sip:6172531000@mit.edu>;tag=4561c4561 To: "Dennis Baron"<sip:6172531000@mit.edu>;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 5 REGISTER Contact: "Dennis Baron"<sip:6172531000@18.10.0.79;LINEID=05523f7a97b54dfa3f0c0e3746d73a24> Expires: 3600 Date: Thu, 30 Sep 2004 00:46:53 GMT Accept-Language: en Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (WinNT) Content-Length: 0 Via: SIP/2.0/UDP 18.10.0.79

  19. SIP REGISTER – 401 Response SIP/2.0 401 Unauthorized From: "Dennis Baron"<sip:6172531000@mit.edu>;tag=4561c4561 To: "Dennis Baron"<sip:6172531000@mit.edu>;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 5 REGISTER Via: SIP/2.0/UDP 18.10.0.79 Www-Authenticate: Digest realm="mit.edu", nonce="f83234924b8ae841b9b0ae8a92dcf0b71096505216", opaque="reg:change4" Date: Thu, 30 Sep 2004 00:46:56 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

  20. SIP REGISTER with Credentials REGISTER sip:mit.edu SIP/2.0 From: "Dennis Baron"<sip:6172531000@mit.edu>;tag=4561c4561 To: "Dennis Baron"<sip:6172531000@mit.edu>;tag=324591026 Call-Id: 9ce902bd23b070ae0108b225b94ac7fa Cseq: 6 REGISTER Contact: "Dennis Baron"<sip:61725231000@18.10.0.79;LINEID=05523f7a97b54dfa3f0c0e3746d73a24> Expires: 3600 Date: Thu, 30 Sep 2004 00:46:53 GMT Accept-Language: en Supported: sip-cc, sip-cc-01, timer, replaces User-Agent: Pingtel/2.1.11 (WinNT) Content-Length: 0 Authorization: DIGEST USERNAME="6172531000@mit.edu", REALM="mit.edu", NONCE="f83234924b8ae841b9b0ae8a92dcf0b71096505216", URI="sip:mit.edu", RESPONSE="ae064221a50668eaad1ff2741fa8df7d", OPAQUE="reg:change4" Via: SIP/2.0/UDP 18.10.0.79

  21. Proxy User B MIT.EDU “Calls” dbaron @MIT.EDU 180 - Ringing 180 - Ringing 100 - Trying 200 - OK INVITE: sip:dbaron@MIT.EDU 200 - OK INVITE: sip:dbaron@18.18.2.4 Rings Answers ACK RTP Talking Talking Hangs up BYE 200 - OK SIP Flows – Via Proxy User A

  22. Proxy Gateway User A 30161 MIT.EDU “Calls” joe @MIT.EDU INVITE: sip:38400@18.162.0.25 BYE 200 - OK INVITE: sip:joe@MIT.EDU 100 - Trying ACK ACK 180 - Ringing 200 - OK 180 - Ringing 200 - OK 200 - OK BYE Rings Answers RTP Talking Talking Hangs up SIP Flows – Via Gateway

  23. SIP INVITE with Record-Route INVITE sip:37669@18.162.0.25 SIP/2.0 Record-Route: <sip:18.7.21.118:5080;lr;a;t=2c41;s=b07e28aa8f94660e8545313a44b9ed50> From: \"Dennis Baron\"<sip:6172531000@mit.edu>;tag=2c41 To: sip:37669@mit.edu Call-Id: call-1096505069-3@18.10.0.79 Cseq: 1 INVITE Contact: \"Dennis Baron\"<sip:6172531000@18.10.0.79> 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/2.1.11 (WinNT) Date: Thu, 30 Sep 2004 00:44:30 GMT Via: SIP/2.0/UDP 18.7.21.118:5080;branch=z9hG4bK2cf12c563cec06fd1849ff799d069cc0 Via: SIP/2.0/UDP 18.7.21.118;branch=z9hG4bKd26e44dfdc2567170d9d32a143a7f4d8 Via: SIP/2.0/UDP 18.10.0.79 Max-Forwards: 17

  24. SIP Standards Just a sampling of IETF standards work… IETF RFCs http://ietf.org/rfc.html • 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

  25. SIP Standards (cont.) • RFC3265 SIP event notification – SUBSCRIBE and NOTIFY • RFC3266 IPv6 support in SDP • RFC3311 SIP UPDATE method – eg. changing media • RFC3325 Asserted identity in trusted networks • RFC3361 Locating outbound SIP proxy with DHCP • RFC3428 SIP extensions for Instant Messaging • RFC3515 SIP REFER method – eg. call transfer • SIMPLE IM/Presence - http://ietf.org/ids.by.wg/simple.html • SIP authenticated identity management - • http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-02.txt

  26. NATs: Hole Punching - Peers tras distinto NAT

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

  28. Terminals • Telephones • Video phones • IVR devices • Voicemail Systems • “Soft phones” (e.g., NetMeeting®)

  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

  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-media-related 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)

  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

  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

  33. The Protocols (cont) • H.323 is a “framework” document that describes how the various pieces fit together • H.225.0 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.225.0 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

  34. The Protocols (cont) • H.450.x is a series of supplementary service protocols • H.460.x is a series of version-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.150.1 defines how to relay modem signals • H.235 defines security within H.323 systems • X.680 defines 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

  35. Registration, Admission, and Status - RAS • Defined in H.225.0 • 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

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

  37. The H323 stack

  38. H323 Clients

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

  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 desarrollada para Linux pero actualmente 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.

  41. ASTERISK • 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 • APIs para desarrollo de nuevos servicios y aplicaciones • Integración con Bases de Datos • Integración con Aplicaciones ya desarrolladas • Código Abierto: sw libre

  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 Internet. • 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.

  43. Configuración básica

  44. Configuración básica (2)

  45. Configuración básica (3)

  46. IMPLEMENTACIÓN DE TELEFONÍA IP EN UNA ORGANIZACIÓN INTEGRACIÓN CISCO-ASTERISK

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

  48. PROTOCOLOS • Sip • H323 • MGCP (Megaco Protocol)

  49. OBJETIVO FINAL

  50. FUNCIONAMIENTO DE CALL MANAGER

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