Post on 10-Feb-2017
TECNOLOGÍAS DE RED AVANZADAS – Master IC 2011-2012 – http://www.grc.upv.es/docencia/tra/
2 -Voz sobre IP (VoIP)
SIP y H.323: Establecimiento y gestión de sesiones multimedia
Asterisk
Thanks to :RADCOM technologiesH. ShulzrinnePaul. E. Jones (from packetizer.com)
Computer Networking: A Top Down Approach
Featuring the Internet, 3rd edition.
Jim Kurose, Keith RossAddison-Wesley, July
2004.
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2 Voice-over-Data (VoD) Enables New Applications
“Click to talk” web sites for e-commerceDigital white-board conferencesBroadcast audio and video over the Internet or a
corporate Intranet Integrated messaging: check (or leave) voice
mail over the Internet Instant messaging
Voicemail notificationsStock notificationsCallback notification
Fax over IPEtc.
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Sesion Initiation Protocol
SIP is end-to-end, client-server session signaling protocolSIP’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 secretaryReply with a webpage if unavailableSend a JPEG on invitation
Features:Textual encoding (telnet, tcpdump compatible).Programmability.Post-dial delay: 1.5 RTTUses either UDP or TCPMulticast/Unicast comm. support3
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Where’s SIP
Application
Transport
Network
Physical/Data Link
4
Ethernet
IP
TCP UDP
RTSP SIP
SDP codecs
RTP DNS(SRV)
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2 IP SIP Phones and Adaptors
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4
1
3
Analog phone adaptor
Palmcontrol
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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)
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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
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SIP Trapezoid
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DNS Server
Location Server
Terminating User Agent
Outgoing Proxy
Originating User Agent
DNS
SIP
SIP
SIP SIP
RTP
Registrar
Incoming Proxy
SIP
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SIP Triangle?
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DNS Server
Location Server
Terminating User Agent
Originating User Agent
DNS
SIP
SIP SIP
RTP
Registrar
Incoming Proxy
SIP
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SIP Peer to Peer!
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Terminating User Agent
Originating User Agent
SIP
RTP
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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 server10
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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 Decline11
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SIP Flows - Basic
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ACK
200 - OK
INVITE: sip:18.18.2.4“Calls” 18.18.2.4
180 - Ringing Rings
200 - OK Answers
BYEHangs up
RTPTalking Talking
User A
User B
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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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CODECs
GIPS Enhanced G.7118kHz sampling rateVoice Activity DetectionVariable bit rate
G.7118kHz sampling rate64kbps
G.7298kHz sampling rate8kbpsVoice Activity Detection
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SIP Flows - Registration
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200 - OK
REGISTER: sip:dbaron@MIT.EDU
401 - Unauthorized
User B MIT.EDUMIT.EDU
Registrar
REGISTER: (add credentials)
MIT.EDUMIT.EDU
Location
sip:dbaron@MIT.EDU
Contact 18.18.2.4
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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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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
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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
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SIP Flows – Via Proxy
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INVITE: sip:dbaron@MIT.EDU“Calls” dbaron @MIT.EDU
INVITE: sip:dbaron@18.18.2.4
100 - Trying
180 - Ringing
Rings180 - Ringing
200 - OK Answers
200 - OK
ACK
BYEHangs up
200 - OK
User A
User BMIT.EDUMIT.EDU
Proxy
Talking TalkingRTP
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SIP Flows – Via Gateway
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INVITE: sip:joe@MIT.EDU“Calls” joe @MIT.EDU
INVITE: sip:38400@18.162.0.25
100 - Trying
ACKACK
User A MIT.EDUMIT.EDU
Proxy30161
Gateway
180 - Ringing
180 - Ringing
Rings
200 - OK
200 - OK
Answers
BYEHangs upBYE
200 - OK
200 - OK
Talking TalkingRTP
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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: 1723
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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 ProtocolRFC1889 RTP - Real-time Transport ProtocolRFC2326 RTSP - Real-Time Streaming ProtocolRFC3262 SIP PRACK method – reliability for 1XX
messagesRFC3263 Locating SIP servers – SRV and NAPTRRFC3264 Offer/answer model for SDP use with
SIP24
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SIP Standards (cont.)
RFC3265 SIP event notification – SUBSCRIBE and NOTIFY
RFC3266 IPv6 support in SDPRFC3311 SIP UPDATE method – eg. changing
mediaRFC3325 Asserted identity in trusted networksRFC3361 Locating outbound SIP proxy with DHCPRFC3428 SIP extensions for Instant MessagingRFC3515 SIP REFER method – eg. call transferSIMPLE IM/Presence -
http://ietf.org/ids.by.wg/simple.htmlSIP authenticated identity management -
http://www.ietf.org/internet-drafts/draft-ietf-sip-identity-02.txt2
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NATs: Hole Punching - Peers tras distinto NAT
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Elements of an H.323 System
TerminalsMultipoint Control Units (MCUs)GatewaysGatekeeperBorder Elements
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Referred to as “endpoints”
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Terminals
TelephonesVideo phones IVR devicesVoicemail 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-media-related functions
The MG handles the mediaGateways 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 follow-me/find-me and forward on busy
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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
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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 conferencingT.38 defines how to relay fax signalsV.150.1 defines how to relay modem signalsH.235 defines security within H.323 systemsX.680 defines the ASN.1 syntax used by the
RecommendationsX.691 defines the Packed Encoding Rules (PER)
used to encode messages for transmission on the network3
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Registration, Admission, and Status - RAS
Defined in H.225.0Allows an endpoint to request authorization to
place or accept a callAllows a Gatekeeper to control access to and
from devices under its controlAllows a Gatekeeper to communicate the address
of other endpointsAllows two Gatekeepers to easily exchange
addressing information
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T GKRRQRCF
ARQ
(endpoint is registered)
ACF(endpoint may place call)
DRQ
DCF(call has terminated) T Terminal
GK Gatekeeper
GW Gateway
Symbol Key:
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The H323 stack
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H323 Clients
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O.S. Client Price
Windows NetMeeting +/- free
Unix (Linux) DC-Share nv
Sun Sunforum +/- free
… ... ... ... ... ...
TECNOLOGÍAS DE RED AVANZADAS – Master IC 2011-2012 – http://www.grc.upv.es/docencia/tra/
2- Voz sobre IP (VoIP)
SIP y H.323: Establecimiento y gestión de sesiones multimedia
Asterisk
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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.4
0
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ASTERISK
Soporta SIP, H.323, MGCP, IAXSe obtiene de : ftp://ftp.digium.com Integra casi todos los codecs de audioSoporte de Telefonía TradicionalSoporte de Telefonía por Voz IPAPIs para desarrollo de nuevos servicios y
aplicaciones Integración con Bases de Datos Integración con Aplicaciones ya desarrolladasCódigo Abierto: sw libre
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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.
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Configuración básica
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Configuración básica (2)
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Configuración básica (3)
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IMPLEMENTACIÓN DE TELEFONÍA IP EN UNA ORGANIZACIÓN
INTEGRACIÓN CISCO-ASTERISK
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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
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PROTOCOLOS
SipH323MGCP (Megaco Protocol)
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OBJETIVO FINAL
49
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8TUV
9WXYZPQRS
* 0OPER
#
7 96 0CI SCO IP P HO NE
ime ssage s di rector ies
setting sser vice s
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8TUV
9WXYZPQRS
* 0OPER
#
79 60CISCO IP P HONE
ime ssage s d ir ector ie s
set tin gsser vic es
CALL MANAGER158.42.250.141
GW KISIN158.42.255.237
CENTRALITA TELÉFONOS
MD-110
GW ALCOI
GW GANDIA
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8TUV
9WXY ZPQRS
* 0OPER
#
7 96 0CI SCO I P PHONE
ime ssages dire ct or ie s
se ttingsser vices
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8TUV
9WXYZPQRS
* 0OPER
#
79 60CIS CO IP PHONE
ime ssa ges dire cto r ies
set tin gsserv ic es
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8TUV
9WXYZPQRS
* 0OPER
#
79 6 0CISCO IP P HONE
ime ssa ges di rectories
set ti ng sser vices
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8T UV
9WXY ZPQRS
* 0OPER
#
79 60CI SCO I P PHONE
im essages direc to ries
setting sse rvices
CAMPUS VALENCIA
CAMPUS ALCOI
CAMPUS GANDÍA
CENTRALITA TELÉFONOS
CENTRALITA TELÉFONOS
ASTERISK158.42.250.173
TECN
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FUNCIONAMIENTO DE CALL MANAGER
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CONFIGURACIÓN CM
Interfaz Webhttps://xxxxxx/CCMAdmin/Main.asp
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TECN
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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
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TECN
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Partitions
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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
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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 patternSe pueden usan caracteres especiales (x…) para
hacer rangos, etcDefinir route patterns para diferentes tipos de
llamadas: nacionales, sin salida….
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ESQUEMA DE NUMERACIÓN
67xxx: Teléfonos IP HW (Vera)68xxx: SoftPhones69xxx: Teléfonos SIP7xxxx: Teléfonos analógicos (fuera del Call
Manager)11xxx: Teléfonos móviles
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Route patterns
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GATEWAYS
Debe haber uno por cada campusOtro que será el router de salida general. Coste: 3500-4000 euros
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Gateways
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TRUNK CON ASTERISK
Es un enlace desde el Call Manager al Asterisk: se enrutan llamadas de uno al otroSe define mediante la IP del Asterisk
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1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8T UV
9WXYZPQRS
* 0OPER
#
79 60CIS CO IP PHONE
imes sa ges direct or ies
sett in gsser vice s
1 2ABC
3DEF
4 5JKL
6MN OGHI
7 8TUV
9WXY ZPQRS
* 0OPER
#
7 96 0CI SC O I P PH ONE
imessag es d ire ctor ie s
se tting sser vices
CALL MANAGER158.42.250.141
GW KISIN158.42.255.237
CENTRALITA TELÉFONOS
MD-110
GW ALCOI
GW GANDIA
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8T UV
9WXYZPQRS
* 0OPER
#
79 60CIS CO I P PHONE
imessa ges di rector ies
set tingsser vices
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8T UV
9WXY ZPQRS
* 0OPER
#
7 960CIS CO IP PHONE
ime ss age s direc tories
se tting sser vices
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8TUV
9WXY ZPQRS
* 0OPER
#
79 6 0CISCO I P PHONE
imessag es dir e ctories
se tt ing sse rvices
1 2ABC
3DEF
4 5JKL
6MNOGHI
7 8TUV
9WXYZPQRS
* 0OPER
#
79 60CIS CO IP P HONE
im essage s dir ector ies
sett ingsser vices
CAMPUS VALENCIA
CAMPUS ALCOI
CAMPUS GANDÍA
CENTRALITA TELÉFONOS
CENTRALITA TELÉFONOS
ASTERISK158.42.250.173
TECN
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Trunk
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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
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Teléfonos
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Teléfonos II
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Teléfonos III
65
Teléfono Cisco Teléfono SIP
300 Euros 45-50 Euros
Configuración desde el CM http://x.y.z.w:9999/
SIP_ADDITIONAL.CONF
TECN
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Teléfonos IV [69001] <--------- Extensión username=69001 <--------- Podría ser el login type=friend record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never mailbox=666@testmail <------ Su buzón 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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Teléfonos V
67
Softphone Cisco
IP Communicator