TECNOLOGAS 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
gestin de sesiones multimedia
Asterisk
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from packetizer.com)
Computer Networking: A
Top Down Approach
Featuring the Internet,
3
rd
edition.
Jim Kurose, Keith Ross
Addison-Wesley, July 2004.

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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.
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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
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Wheres SIP




Application

Transport

Network

Physical/Data Link


4
Ethernet
IP
TCP UDP
RTSP SIP
SDP codecs
RTP DNS(SRV)
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IP SIP Phones and Adaptors
5
4
1
3
Analog phone adaptor
Palm
control
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4
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
7
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?
8
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!
9
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 users address to server
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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 Decline
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SIP Flows - Basic
1
2
ACK
200 - OK
INVITE: sip:18.18.2.4
Calls
18.18.2.4
180 - Ringing Rings
200 - OK Answers
BYE Hangs up
RTP Talking Talking
User
A
User
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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.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

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SIP Flows - Registration
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200 - OK
REGISTER: sip:dbaron@MIT.EDU
401 - Unauthorized
User
B
MIT.EDU
Registrar
REGISTER: (add credentials)
MIT.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
2
1
INVITE: sip:dbaron@MIT.EDU
Calls dbaron
@MIT.EDU
INVITE: sip:dbaron@18.18.2.4
100 - Trying
180 - Ringing
Rings 180 - Ringing
200 - OK Answers
200 - OK
ACK
BYE Hangs up
200 - OK
User
A
User
B
MIT.EDU
Proxy
Talking Talking RTP
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SIP Flows Via Gateway
2
2
INVITE: sip:joe@MIT.EDU
Calls joe
@MIT.EDU
INVITE: sip:38400@18.162.0.25
100 - Trying
ACK
ACK
User
A
MIT.EDU
Proxy
30161
Gateway
180 - Ringing
180 - Ringing
Rings
200 - OK
200 - OK
Answers
BYE Hangs up
BYE
200 - OK
200 - OK
Talking Talking RTP
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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: 17
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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 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-sip-
identity-02.txt
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2
6

NATs: Hole Punching - Peers tras distinto NAT
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Elements of an H.323 System
Terminals
Multipoint Control Units (MCUs)
Gateways
Gatekeeper
Border Elements

2
7
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-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)
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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
3
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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 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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Registration, Admission, and Status RAS (cont)
3
6
T GK
RRQ
RCF
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
3
8
O.S. Client Price
Windows NetMeeting +/- free
Unix (Linux) DC-Share nv
Sun Sunforum +/- free
... ... ... ... ...
TECNOLOGAS 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
gestin de sesiones multimedia
Asterisk
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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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Configuracin bsica (2)

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Configuracin bsica (3)
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IMPLEMENTACIN DE TELEFONA IP
EN UNA ORGANIZACIN

INTEGRACIN CISCO-ASTERISK
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CARACTERISTICAS CISCO CALL MANAGER
Solucin de Telefona IP de Cisco
Distribuible
Escalable (30000 lineas/servidor)
Soporta muchos usuarios
Sobre Windows o linux
Soporta gran variedad de telfonos

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PROTOCOLOS
Sip
H323
MGCP (Megaco Protocol)

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OBJETIVO FINAL
4
9
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
CALL MANAGER
158.42.250.141
GW KISIN
158.42.255.237
CENTRALITA
TELFONOS
MD-110
GW ALCOI
GW GANDIA
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
CAMPUS VALENCIA
CAMPUS ALCOI
CAMPUS GANDA
CENTRALITA
TELFONOS
CENTRALITA
TELFONOS
ASTERISK
158.42.250.173
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FUNCIONAMIENTO DE CALL MANAGER
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CONFIGURACIN CM
Interfaz Web
https://xxxxxx/CCMAdmin/Main.asp
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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
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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: Telfonos IP HW (Vera)
68xxx: SoftPhones
69xxx: Telfonos SIP
7xxxx: Telfonos analgicos (fuera del Call Manager)
11xxx: Telfonos mviles
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Route patterns
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GATEWAYS
Debe haber uno por cada campus
Otro que ser 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 otro
Se define mediante
la IP del Asterisk
6
0
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
CALL MANAGER
158.42.250.141
GW KISIN
158.42.255.237
CENTRALITA
TELFONOS
MD-110
GW ALCOI
GW GANDIA
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services 1 2 ABC 3 DEF
4 5 JKL 6 MNO GHI
7 8 TUV 9 WXYZ PQRS
* 0 OPER #
7960 CISCO IP PHONE
i messages directories
settings services
CAMPUS VALENCIA
CAMPUS ALCOI
CAMPUS GANDA
CENTRALITA
TELFONOS
CENTRALITA
TELFONOS
ASTERISK
158.42.250.173
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Trunk
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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
6
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Telfonos
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Telfonos II
6
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Telfonos III
6
5







Telfono Cisco Telfono SIP
300 Euros 45-50 Euros
Configuracin desde el CM 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
6
6
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Telfonos V
6
7
Softphone Cisco
IP Communicator