Voip

2-
Voz sobre IP (VoIP)
SIP y H.323: Establecimiento y

gestin de sesiones multimedia
Computer Networking: A
Asterisk Top Down Approach
Featuring the Internet,
3rd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July
2004.
Thanks to :
RADCOM technologies
H. Shulzrinne
Paul. E. Jones (from
packetizer.com)
TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

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.
2
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
3
Multicast/Unicast comm. support
Wheres SIP
SDP codecs
Application RTSP SIP RTP DNS(SRV)
Transport TCP UDP
Network IP
Physical/Data Link Ethernet
4
IP SIP Phones and Adaptors
2 1
Aretrue Internet
hosts
Choice of Analog phone
application adaptor
Choice of server
3
IP appliances
Implementations
Palm
3Com (3) contro
l
Columbia
University
4
MCI WorldCom (2)
Mediatrix (1)
5 Nortel (4) 5
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
A Sip Gateway service allows you to call 'real' numbers from your software or
6 have a dedicated 'real' telephone number which comes in via VoIP
SIP Trapezoid
DNS Location
Server Server
DNS
Registrar
SIP
Outgoing Incoming
Proxy Proxy
SIP SIP SIP
SIP
Originating Terminating
User Agent RTP User Agent
7
SIP Triangle?
DNS Location
Server Server
DNS
Registrar
Incoming
Proxy
SIP SIP SIP
SIP
8
SIP Peer to Peer!
SIP
9
SIP Methods
INVITE Requests a session
ACK Final response to the INVITE
OPTIONSAsk for server capabilities
CANCEL Cancels a pending request
BYE Terminates a session
REGISTER Sends users address to server

1
0
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

1
1
SIP Flows - Basic
User User
A B
Calls
INVITE: sip:18.18.2.4
18.18.2.4
180 - Ringing Rings
200 - OK Answers
ACK
Talking RTP Talking
Hangs up BYE
200 - OK
1
2
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
1
3
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)
1
4
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
1
5
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
1
6
SIP Flows - Registration
User Registrar Location

B MIT.EDU MIT.EDU
REGISTER: sip:dbaron@MIT.EDU
401 - Unauthorized
REGISTER: (add credentials)

sip:dbaron@MIT.EDU
Contact 18.18.2.4
200 - OK
1
7
SIP REGISTER
REGISTER sip:mit.edu SIP/2.0

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
Content-Length: 0
Via: SIP/2.0/UDP 18.10.0.79
1
8
SIP REGISTER 401 Response
SIP/2.0 401 Unauthorized

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
1
9
SIP REGISTER with Credentials
REGISTER sip:mit.edu SIP/2.0

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
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
2
0
SIP Flows Via Proxy
User Proxy User

A MIT.EDU B
Calls dbaron
INVITE: sip:dbaron@MIT.EDU
@MIT.EDU
INVITE: sip:dbaron@18.18.2.4
100 - Trying
180 - Ringing Rings
180 - Ringing
200 - OK Answers
200 - OK
ACK
Talking RTP Talking
Hangs up BYE
200 - OK
2
1
SIP Flows Via Gateway
User Proxy Gateway

A MIT.EDU 30161
Calls joe
INVITE: sip:joe@MIT.EDU
@MIT.EDU
INVITE: sip:38400@18.162.0.25
100 - Trying Rings
180 - Ringing
180 - Ringing
Answers
200 - OK
200 - OK
ACK
ACK
Talking RTP Talking
Hangs up BYE
BYE
200 - OK
200 - OK
2
2
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
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
2
3
SIP Standards
Just a sampling of IETF standards work

IETF RFCs http://ietf.org/rfc.html
RFC3261Core SIP specification obsoletes
RFC2543
RFC2327SDP Session Description Protocol
RFC1889RTP - Real-time Transport Protocol
RFC2326RTSP - Real-Time Streaming Protocol
RFC3262SIP PRACK method reliability for 1XX
messages
RFC3263Locating SIP servers SRV and NAPTR
RFC3264Offer/answer model for SDP use with SIP
2
4
SIP Standards (cont.)
RFC3265SIP event notification SUBSCRIBE and

NOTIFY
RFC3266IPv6 support in SDP
RFC3311SIP UPDATE method eg. changing media
RFC3325Asserted identity in trusted networks
RFC3361Locating outbound SIP proxy with DHCP
RFC3428SIP extensions for Instant Messaging
RFC3515SIP 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
2
5
6
2
NATs: Hole Punching - Peers tras distinto NAT

Elements of an H.323 System
Terminals Referred to as
Multipoint Control Units (MCUs)endpoints
Gateways
Gatekeeper
Border Elements
2
7
Terminals
Telephones
Video phones
IVR devices
Voicemail Systems
Soft phones (e.g., NetMeeting)
2
8
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
2
9
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)
3
0
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
1
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
3
2
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
3
3
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
3 network
4
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
3
5
Registration, Admission, and Status RAS
(cont)
T RRQ GK
RCF
(endpoint is registered)
ARQ
ACF
(endpoint may place call)
DRQ Symbol Key:
(call has terminated) T Terminal
DCF GK Gatekeeper
GW Gateway
3
6
7
3
The H323 stack

H323 Clients
O.S. Client Price
Windows NetMeeting +/- free
Unix (Linux) DC-Share nv
Sun Sunforum +/- free
... ... ... ... ...
3
8
2-
Voz sobre IP (VoIP)
SIP y H.323: Establecimiento y

gestin de sesiones multimedia
Asterisk

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
4 ms sencilla de Asterisk.
0
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
4
1
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.
4
2
3
4
Configuracin bsica
4
4
Configuracin bsica (2)

5
4
Configuracin bsica (3)

IMPLEMENTACIN DE TELEFONA
IP EN UNA ORGANIZACIN
INTEGRACIN CISCO-ASTERISK
4
6
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
4
7
8
4
Sip
H323
MGCP (Megaco Protocol)
PROTOCOLOS
9
4
OBJETIVO FINAL
0
5
FUNCIONAMIENTO DE CALL MANAGER

CONFIGURACIN CM
Interfaz Web
https://xxxxxx/CCMAdmin/Main.asp
5
1
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
5
2
3
5
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
5
4
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.
5
5
ESQUEMA DE NUMERACIN
67xxx: Telfonos IP HW (Vera)

68xxx: SoftPhones
69xxx: Telfonos SIP
7xxxx: Telfonos analgicos (fuera del Call
Manager)
11xxx: Telfonos mviles
5
6
7
5
Route patterns
GATEWAYS
Debe haber uno por cada campus

Otro que ser el router de salida general.
Coste: 3500-4000 euros
5
8
9
5
Gateways
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
6
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
6
2
3
6
Telfonos
4
6
Telfonos II
Telfonos III
Telfono Cisco Telfono SIP

300 Euros 45-50 Euros
Configuracin desde el CM http://x.y.z.w:9999/
SIP_ADDITIONAL.CONF
6
5
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
7
6
Softphone Cisco
IP Communicator
Telfonos V

Voip

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2-

Voz sobre IP (VoIP)

SIP y H.323: Establecimiento y

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

Sesion Initiation Protocol

SIP is end-to-end, client-server session signaling

Application RTSP SIP RTP DNS(SRV)

Transport TCP UDP

Physical/Data Link Ethernet

IP SIP Phones and Adaptors

SIP Peer to Peer!

INVITE Requests a session

ACK Final response to the INVITE

OPTIONSAsk for server capabilities

CANCEL Cancels a pending request

BYE Terminates a session

REGISTER Sends users address to server

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

SIP Flows - Basic

180 - Ringing Rings

Talking RTP Talking

INVITE sip:e9-airport.mit.edu SIP/2.0

From: "Dennis Baron"<sip:6172531000@mit.edu>;tag=1c41

Contact: "Dennis Baron"<sip:6172531000@18.10.0.79>

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

Session Description Protocol

IETF RFC 2327

o=Pingtel 5 5 IN IP4 18.10.0.79

c=IN IP4 18.10.0.79

m=audio 8766 RTP/AVP 96 97 0 8 18 98

GIPS Enhanced G.711

SIP Flows - Registration

User Registrar Location

REGISTER: (add credentials)

REGISTER sip:mit.edu SIP/2.0

SIP REGISTER 401 Response

SIP/2.0 401 Unauthorized

SIP REGISTER with Credentials

REGISTER sip:mit.edu SIP/2.0

SIP Flows Via Proxy

User Proxy User

Talking RTP Talking

SIP Flows Via Gateway

User Proxy Gateway

Talking RTP Talking

SIP INVITE with Record-Route

INVITE sip:37669@18.162.0.25 SIP/2.0

Just a sampling of IETF standards work

SIP Standards (cont.)

RFC3265SIP event notification SUBSCRIBE and

NATs: Hole Punching - Peers tras distinto NAT

Elements of an H.323 System

Responsible for managing multipoint conferences

The Gateway is composed of a Media Gateway

The Gatekeeper is an optional component in the