Voip

2Voz sobre IP (VoIP)
SIP y H.323: Establecimiento y gestin de sesiones multimedia Asterisk
3rd edition. Jim Kurose, Keith Ross Addison-Wesley, July 2004.
Computer Networking: A Top Down Approach Featuring the Internet,
Thanks to : RADCOM technologies H. Shulzrinne Paul. E. Jones (from packetizer.com)
TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012 http://www.grc.upv.es/docencia/tra/
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 Multicast/Unicast comm. support
Wheres SIP
SDP
codecs
Application
RTSP
SIP
RTP
DNS(SRV)
Transport
TCP
UDP
Network
IP
Physical/Data Link
Ethernet
IP SIP Phones and Adaptors 2

Are
true Internet hosts
Choice of application Choice of server IP appliances
Analog phone adaptor 3

Palm control
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
6
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
Location Server Registrar Incoming Proxy

SIP SIP
Outgoing Proxy
SIP
Originating User Agent
SIP RTP
Terminating User Agent
SIP Triangle?
DNS Server
DNS
Location Server Registrar Incoming Proxy

SIP SIP SIP
SIP RTP
SIP Peer to Peer!
SIP RTP
SIP Methods
INVITE
Requests a session
ACK
OPTIONS CANCEL BYE REGISTER
Final response to the INVITE

Ask for server capabilities Cancels a pending request Terminates a session Sends users address to server
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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
404 Not Found

504 Server Time-out 603 Decline
1 1
SIP Flows - Basic

User A User B
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
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
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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

User B Registrar MIT.EDU Location 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 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
1 9
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
2 0
SIP Flows Via Proxy

User A Proxy MIT.EDU User B
Calls dbaron @MIT.EDU
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
2 1
SIP Flows Via Gateway

User A Proxy MIT.EDU Gateway 30161
Calls joe @MIT.EDU
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
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 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
2 3
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
2 4
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
2 5
2 6
NATs: Hole Punching - Peers tras distinto NAT
Elements of an H.323 System Terminals Multipoint Control Units (MCUs) Gateways Gatekeeper Border Elements
Referred to as endpoints
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-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)
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 followme/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 network
3 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)
RRQ RCF
(endpoint is registered)
GK
ARQ ACF
(endpoint may place call)
DRQ
(call has terminated)
Symbol Key: T GK GW Terminal Gatekeeper Gateway
DCF
3 6
3 7
The H323 stack
H323 Clients
O.S. Windows Unix (Linux)
Client NetMeeting DC-Share
Price +/- free nv
Sun
...
Sunforum
... ...
+/- free
... ...
3 8
2Voz sobre IP (VoIP)

SIP y H.323: Establecimiento y gestin de sesiones multimedia Asterisk
TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012 http://www.grc.upv.es/docencia/tra/
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.
4 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
4 3
Configuracin bsica
4 4
Configuracin bsica (2)
4 5
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
4 8
Sip H323 MGCP (Megaco Protocol) PROTOCOLOS
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
GW ALCOI CALL MANAGER

158.42.250.141 CENTRALITA TELFONOS
ASTERISK
158.42.250.173
CAMPUS GANDA
GW KISIN
158.42.255.237
CENTRALITA TELFONOS
CENTRALITA TELFONOS MD-110
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
4 9
5 0
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
5 3
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: 68xxx: 69xxx: 7xxxx: 11xxx: Telfonos IP HW (Vera) SoftPhones Telfonos SIP Telfonos analgicos (fuera del Call Manager) Telfonos mviles
5 6
5 7
Route patterns
GATEWAYS Debe haber uno por cada campus Otro que ser el router de salida general. Coste: 3500-4000 euros
5 8
5 9
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
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
GW ALCOI CALL MANAGER

158.42.250.141 CENTRALITA TELFONOS
ASTERISK
158.42.250.173
CAMPUS GANDA
GW KISIN
158.42.255.237
CENTRALITA TELFONOS
CENTRALITA TELFONOS MD-110
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
6 1
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
6 3
Telfonos
6 4
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
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
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Softphone Cisco
IP Communicator
Telfonos V

Voip

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2Voz sobre IP (VoIP)

SIP y H.323: Establecimiento y gestin de sesiones multimedia Asterisk

3rd edition. Jim Kurose, Keith Ross Addison-Wesley, July 2004.

Computer Networking: A Top Down Approach Featuring the Internet,

Thanks to : RADCOM technologies H. Shulzrinne Paul. E. Jones (from packetizer.com)

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012 http://www.grc.upv.es/docencia/tra/

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

Fax over IP Etc.

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

Sesion Initiation Protocol SIP is end-to-end, client-server session signaling protocol

Arbitrary services built on top of SIP, e.g.:

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

IP SIP Phones and Adaptors 2

true Internet hosts

Choice of application Choice of server IP appliances

Analog phone adaptor 3

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

Location Server Registrar Incoming Proxy

Originating User Agent

Terminating User Agent

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

Location Server Registrar Incoming Proxy

Originating User Agent

Terminating User Agent

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

SIP Peer to Peer!

Originating User Agent

Terminating User Agent

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

Final response to the INVITE

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

404 Not Found

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

SIP Flows - Basic

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

CODECs GIPS Enhanced G.711

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

SIP Flows - Registration

REGISTER: (add credentials)

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

SIP REGISTER 401 Response

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

SIP REGISTER with Credentials

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

SIP Flows Via Proxy

Calls dbaron @MIT.EDU

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

SIP Flows Via Gateway

Calls joe @MIT.EDU

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

SIP INVITE with Record-Route

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012

NATs: Hole Punching - Peers tras distinto NAT

TECNOLOGAS DE RED AVANZADAS Master IC 2011-2012