Tema 5: Voz sobre IP VoIP

1
Tema 5 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
• SIPs 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
Wheres SIP

• Application
• Transport
• Network
• Physical/Data Link

SDP
codecs
RTSP
SIP
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
DNS Server
Location Server
DNS
Registrar
SIP
Outgoing Proxy
Incoming Proxy
SIP
SIP
SIP
SIP
Originating User Agent
Terminating User Agent
RTP
8
SIP Triangle?
DNS Server
Location 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 users 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
SIP Flows - Basic
13
SIP INVITE

• INVITE sipe9-airport.mit.edu SIP/2.0
• From "Dennis Baron"tag1
  c41
• To sipe9-airport.mit.edu
• Call-Id call-1096504121-2_at_18.10.0.79
• Cseq 1 INVITE
• Contact "Dennis Baron"
  
• 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 002842 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

• v0
• oPingtel 5 5 IN IP4 18.10.0.79
• sphone-call
• cIN IP4 18.10.0.79
• t0 0
• maudio 8766 RTP/AVP 96 97 0 8 18 98
• artpmap96 eg711u/8000/1
• artpmap97 eg711a/8000/1
• artpmap0 pcmu/8000/1
• artpmap8 pcma/8000/1
• artpmap18 g729/8000/1
• afmtp18 annexbno
• artpmap98 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
SIP Flows - Registration
18
SIP REGISTER

• REGISTER sipmit.edu SIP/2.0
• From "Dennis Baron"tag4
  561c4561
• To "Dennis Baron"tag324
  591026
• Call-Id 9ce902bd23b070ae0108b225b94ac7fa
• Cseq 5 REGISTER
• Contact "Dennis Baron"LINEID05523f7a97b54dfa3f0c0e3746d73a24
• Expires 3600
• Date Thu, 30 Sep 2004 004653 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"tag4
  561c4561
• To "Dennis Baron"tag324
  591026
• Call-Id 9ce902bd23b070ae0108b225b94ac7fa
• Cseq 5 REGISTER
• Via SIP/2.0/UDP 18.10.0.79
• Www-Authenticate Digest realm"mit.edu",
  nonce"f83234924b8ae841b9b0ae8a92dcf0b71096505216"
  , opaque"regchange4"
• Date Thu, 30 Sep 2004 004656 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 sipmit.edu SIP/2.0
• From "Dennis Baron"tag4
  561c4561
• To "Dennis Baron"tag324
  591026
• Call-Id 9ce902bd23b070ae0108b225b94ac7fa
• Cseq 6 REGISTER
• Contact "Dennis Baron"LINEID05523f7a97b54dfa3f0c0e3746d73a24
• Expires 3600
• Date Thu, 30 Sep 2004 004653 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_at_mit.edu
  ", REALM"mit.edu", NONCE"f83234924b8ae841b9b0ae8
  a92dcf0b71096505216", URI"sipmit.edu",
  RESPONSE"ae064221a50668eaad1ff2741fa8df7d",
  OPAQUE"regchange4"
• Via SIP/2.0/UDP 18.10.0.79

21
SIP Flows Via Proxy
User A
22
SIP Flows Via Gateway
23
SIP INVITE with Record-Route

• INVITE sip37669_at_18.162.0.25 SIP/2.0
• Record-Route b07e28aa8f94660e8545313a44b9ed50
• From \"Dennis Baron\"tag
  2c41
• To sip37669_at_mit.edu
• Call-Id call-1096505069-3_at_18.10.0.79
• Cseq 1 INVITE
• Contact \"Dennis Baron\"9
• 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 004430 GMT
• Via SIP/2.0/UDP 18.7.21.1185080branchz9hG4bK2c
  f12c563cec06fd1849ff799d069cc0
• Via SIP/2.0/UDP 18.7.21.118branchz9hG4bKd26e44d
  fdc2567170d9d32a143a7f4d8
• 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/sim
  ple.html
• SIP authenticated identity management -
• http//www.ietf.org/internet-drafts/draft-ie
  tf-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
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
You can find a bigger list at http//www.openh323
.org/h323_clients.html
39
Tema 5 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
51
CONFIGURACIÓN CM

• Interfaz Web
• https//xxxxxx/CCMAdmin/Main.asp

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

53
Partitions
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

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.

56
ESQUEMA DE NUMERACIÓN

• 67xxx Teléfonos IP HW (Vera)
• 68xxx SoftPhones
• 69xxx Teléfonos SIP
• 7xxxx Teléfonos analógicos (fuera del Call
  Manager)
• 11xxx Teléfonos móviles

57
Route patterns
58
GATEWAYS

• Debe haber uno por cada campus
• Otro que será el router de salida general.
• Coste 3500-4000 euros

59
Gateways
60
TRUNK CON ASTERISK

• Es un enlace desde
• el Call Manager
• al Asterisk
• se enrutan llamadas
• de uno al otro
• Se define mediante
• la IP del Asterisk

61
Trunk
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

63
Teléfonos
64
Teléfonos II
65
Teléfonos III
Teléfono Cisco
Teléfono SIP 300 Euros

45-50 Euros Configuración desde el CM
http//x.y.z.w9999/

SIP_ADDITIONAL.CONF
66
Teléfonos IV

• 69001
• username69001
• typefriend
• record_outAdhoc
• record_inAdhoc
• qualifyno
• port5060
• natnever
• mailbox666_at_testmail asociado (en el voicemail.conf)
• hostdynamic
• dtmfmodeinfo
• contextfrom-internal
• canreinviteno
• calleriddevice
• languagees

67
Teléfonos V
Softphone Cisco IP Communicator