Tema 5: Voz sobre IP

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

5

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



DNS

SIP

SIP SIP

RTP

Registrar

Incoming Proxy

SIP

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SIP Peer to Peer!

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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:[email protected]>;tag=1c41

To: sip:e9-airport.mit.edu

Call-Id: [email protected]

Cseq: 1 INVITE

Contact: "Dennis Baron"<sip:[email protected]>

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:[email protected]

401 - Unauthorized

User B MIT.EDUMIT.EDU

Registrar

REGISTER: (add credentials)

MIT.EDUMIT.EDU

Location

sip:[email protected]

Contact 18.18.2.4

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SIP REGISTER

REGISTER sip:mit.edu SIP/2.0


To: "Dennis Baron"<sip:[email protected]>;tag=324591026

Call-Id: 9ce902bd23b070ae0108b225b94ac7fa

Cseq: 5 REGISTER

Contact: "Dennis Baron"<sip:[email protected];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

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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

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SIP REGISTER with Credentials

REGISTER sip:mit.edu SIP/2.0




Cseq: 6 REGISTER

Contact: "Dennis Baron"<sip:[email protected];LINEID=05523f7a97b54dfa3f0c0e3746d73a24>

Expires: 3600

Date: Thu, 30 Sep 2004 00:46:53 GMT

Accept-Language: en



Content-Length: 0

Authorization: DIGEST USERNAME="[email protected]", 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:[email protected]“Calls” dbaron @MIT.EDU

INVITE: sip:[email protected]

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:[email protected]“Calls” joe @MIT.EDU

INVITE: sip:[email protected]

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:[email protected] SIP/2.0

Record-Route: <sip:18.7.21.118:5080;lr;a;t=2c41;s=b07e28aa8f94660e8545313a44b9ed50>

From: \"Dennis Baron\"<sip:[email protected]>;tag=2c41

To: sip:[email protected]

Call-Id: [email protected]

Cseq: 1 INVITE

Contact: \"Dennis Baron\"<sip:[email protected]>

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: 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

http://ietf.org/rfc.html

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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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http://ietf.org/ids.by.wg/simple.html

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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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2 Registration, Admission, and Status – RAS (cont)

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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

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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



ASTERISK158.42.250.173

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FUNCIONAMIENTO DE CALL MANAGER

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CONFIGURACIÓN CM

Interfaz Webhttps://xxxxxx/CCMAdmin/Main.asp

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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 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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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

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* 0OPER

#


imes sa ges direct or ies

sett in gsser vice s

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#

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


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

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7 8T UV

9WXY ZPQRS

* 0OPER

#


ime ss age s direc tories

se tting sser vices

1 2ABC

3DEF

4 5JKL

6MNOGHI

7 8TUV

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* 0OPER

#

79 6 0CISCO I P PHONE

imessag es dir e ctories

se tt ing sse rvices

1 2ABC

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4 5JKL

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7 8TUV

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#

79 60CIS CO IP P HONE

im essage s dir ector ies

sett ingsser vices

CAMPUS VALENCIA

CAMPUS ALCOI

CAMPUS GANDÍA



ASTERISK158.42.250.173

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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

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Teléfono Cisco Teléfono SIP

300 Euros 45-50 Euros

Configuración desde el CM http://x.y.z.w:9999/

SIP_ADDITIONAL.CONF

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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

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Softphone Cisco

IP Communicator

Tema 5: Voz sobre IP

Documents

Transcript of Tema 5: Voz sobre IP