Escolar Documentos
Profissional Documentos
Cultura Documentos
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
Features:
Textual encoding (telnet, tcpdump compatible). Programmability. Post-dial delay: 1.5 RTT Uses either UDP or TCP Multicast/Unicast comm. support
Wheres SIP
SDP
codecs
Application
RTSP
SIP
RTP
DNS(SRV)
Transport
TCP
UDP
Network
IP
Physical/Data Link
Ethernet
Implementations 3Com (3) Columbia University MCI WorldCom (2) Mediatrix (1) 4 Nortel (4) Siemens (5)
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
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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
SIP Trapezoid
DNS Server
DNS SIP
Outgoing Proxy
SIP
SIP RTP
SIP Triangle?
DNS Server
DNS
SIP RTP
SIP RTP
SIP Methods
INVITE
Requests a session
ACK
OPTIONS CANCEL BYE REGISTER
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SIP Responses 1XX 2XX 3XX Provisional Successful Redirection 100 Trying 200 OK 302 Moved Temporarily
4XX
5XX 6XX
Client Error
Server Error Global Failure
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Calls 18.18.2.4
INVITE: sip:18.18.2.4
180 - Ringing
Rings
200 - OK
Answers
ACK
Talking
RTP
Talking
Hangs up
BYE
200 - OK
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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
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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)
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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
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G.711
8kHz sampling rate 64kbps
G.729
8kHz sampling rate 8kbps Voice Activity Detection
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REGISTER: sip:dbaron@MIT.EDU
401 - Unauthorized
sip:dbaron@MIT.EDU
Contact 18.18.2.4 200 - OK
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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
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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
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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
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INVITE: sip:dbaron@MIT.EDU
INVITE: sip:dbaron@18.18.2.4 100 - Trying 180 - Ringing 180 - Ringing Rings
200 - OK 200 - OK
Answers
ACK
Talking
RTP
Talking
Hangs up
BYE 200 - OK
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INVITE: sip:joe@MIT.EDU
INVITE: sip:38400@18.162.0.25 100 - Trying 180 - Ringing 180 - Ringing Answers 200 - OK 200 - OK ACK ACK Rings
Talking
RTP
Talking
Hangs up
BYE BYE
200 - OK 200 - OK
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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
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Max-Forwards: 17
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
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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-sipidentity-02.txt
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Elements of an H.323 System Terminals Multipoint Control Units (MCUs) Gateways Gatekeeper Border Elements
Referred to as endpoints
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Terminals Telephones Video phones IVR devices Voicemail Systems Soft phones (e.g., NetMeeting)
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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
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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)
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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 followme/find-me and forward on busy
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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
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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
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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
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RRQ RCF
(endpoint is registered)
GK
ARQ ACF
(endpoint may place call)
DRQ
(call has terminated)
DCF
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H323 Clients
Sun
...
Sunforum
... ...
+/- free
... ...
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ASTERISK Aplicacin de software libre que implementa los servicios de una centralita telefnica de VoIP. Permite conectar telfonos de VoIP (que tambin pueden ser programas de ordenador o softphones), fax, lneas RDSI, lneas telefnicas analgicas convencionales Inicialmente desarrollada para Linux pero actualmente existen versiones para casi todas las plataformas. trixbox (con t minscula) es una distribucin Linux (en concreto de CentOS) que incluye Asterisk y FreePBX que es un entorno grfico basado en WEB para una configuracin cmoda y ms sencilla de Asterisk.
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ASTERISK Soporta SIP, H.323, MGCP, IAX Se obtiene de : ftp://ftp.digium.com Integra casi todos los codecs de audio Soporte de Telefona Tradicional Soporte de Telefona por Voz IP APIs para desarrollo de nuevos servicios y aplicaciones Integracin con Bases de Datos Integracin con Aplicaciones ya desarrolladas Cdigo Abierto: sw libre
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IAX (Inter-Asterisk eXchange) Actualmente en la versin 2 (IAX2) es un protocolo que aborda el problema de los NATs. Utilizar el mismo puerto UDP para la sealizacin y para la transmisin de los datos (RTP). Simplifica el nmero 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 ser el futuro de VoIP y otros plantean que la regulacin en tema de NATs, o incluso su desaparicin con la entrada de IPv6 dejaran a SIP en su posicin de liderato.
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Configuracin bsica
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OBJETIVO FINAL
CAMPUS ALCOI
CISCO IP PHONE
7960
CISCO IP PHONE
7960
1 4
GHI
2
ABC
3
DEF
messages
directories
1 4
GHI
2
ABC
3
DEF
messages
directories
i
services settings
i
services settings
5
JKL
6
MNO
5
JKL
6
MNO
7
PQRS
8
TUV
9
WXYZ
7
PQRS
8
TUV
9
WXYZ
0
OPER
0
OPER
CAMPUS VALENCIA
CISCO IP PHONE
7960
CISCO IP PHONE
7960
1 4
GHI
2
ABC
3
DEF
messages
directories
1 4
GHI
2
ABC
3
DEF
messages
directories
i
services settings
i
services settings
5
JKL
6
MNO
5
JKL
6
MNO
7
PQRS
8
TUV
9
WXYZ
7
PQRS
8
TUV
9
WXYZ
0
OPER
0
OPER
ASTERISK
158.42.250.173
CAMPUS GANDA
GW KISIN
158.42.255.237
CENTRALITA TELFONOS
GW GANDIA
CISCO IP PHONE
7960
CISCO IP PHONE
7960
1 4
GHI
2
ABC
3
DEF
messages
directories
1 4
GHI
2
ABC
3
DEF
messages
directories
i
services settings
i
services settings
5
JKL
6
MNO
5
JKL
6
MNO
7
PQRS
8
TUV
9
WXYZ
7
PQRS
8
TUV
9
WXYZ
0
OPER
0
OPER
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5 0
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PARTITIONS Dividen el conjunto de route patterns en subconjuntos de destinos alcanzables identificados por un nombre. Una particin contiene una lista de Route Patterns Facilitan el enrutado de llamadas dividiendo el route plan en subconjuntos lgicos que se pueden basar en la organizacin, localizacin y tipo de llamada
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Partitions
SEARCH SPACES
Es una lista ordenada de rutas de particin. Estas rutas se asocian a los dispositivos (telfonos). Determinan las particiones que los dispositivos que hacen una llamada buscan para que esta llamada se realice
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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.
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ESQUEMA DE NUMERACIN 67xxx: 68xxx: 69xxx: 7xxxx: 11xxx: Telfonos IP HW (Vera) SoftPhones Telfonos SIP Telfonos analgicos (fuera del Call Manager) Telfonos mviles
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Route patterns
GATEWAYS Debe haber uno por cada campus Otro que ser el router de salida general. Coste: 3500-4000 euros
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Gateways
Es un enlace desde el Call Manager al Asterisk: se enrutan llamadas de uno al otro Se define mediante la IP del Asterisk
CAMPUS ALCOI
CISCO IP PHONE
7960
CISCO IP PHONE
7960
1 4
GHI
2
ABC
3
DEF
messages
directories
1 4
GHI
2
ABC
3
DEF
messages
directories
i
services settings
i
services settings
5
JKL
6
MNO
5
JKL
6
MNO
7
PQRS
8
TUV
9
WXYZ
7
PQRS
8
TUV
9
WXYZ
0
OPER
0
OPER
CAMPUS VALENCIA
CISCO IP PHONE
7960
CISCO IP PHONE
7960
1 4
GHI
2
ABC
3
DEF
messages
directories
1 4
GHI
2
ABC
3
DEF
messages
directories
i
services settings
i
services settings
5
JKL
6
MNO
5
JKL
6
MNO
7
PQRS
8
TUV
9
WXYZ
7
PQRS
8
TUV
9
WXYZ
0
OPER
0
OPER
ASTERISK
158.42.250.173
CAMPUS GANDA
GW KISIN
158.42.255.237
CENTRALITA TELFONOS
GW GANDIA
6 0
CISCO IP PHONE
7960
CISCO IP PHONE
7960
1 4
GHI
2
ABC
3
DEF
messages
directories
1 4
GHI
2
ABC
3
DEF
messages
directories
i
services settings
i
services settings
5
JKL
6
MNO
5
JKL
6
MNO
7
PQRS
8
TUV
9
WXYZ
7
PQRS
8
TUV
9
WXYZ
0
OPER
0
OPER
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Trunk
TELEFONOS un identificador, el Device Name (3 caracteres ms la direccin MAC ) una descripcin (ej . la persona asociada) el pool al que corresponde su estado (registrado o no) la direccin IP del telfono: slo se muestra si el telfono est registrado
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Telfonos
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Telfonos II
Telfonos III
Telfono Cisco
300 Euros Configuracin desde el CM
Telfono SIP
45-50 Euros http://x.y.z.w:9999/ SIP_ADDITIONAL.CONF
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Telfonos IV
[69001] <--------- Extensin username=69001 <--------- Podra ser el login type=friend record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never mailbox=666@testmail <------ Su buzn de voz asociado (en el voicemail.conf) host=dynamic dtmfmode=info context=from-internal canreinvite=no callerid=device <69001> language=es
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Softphone Cisco
IP Communicator
Telfonos V