Voice over IPVoice over IPo ce o eo ce o e

Voice over IP

RoadmapRoadmapRoadmapRoadmapIntroduzione

IP Telephony/VoIP

Componenti IP TelephonyComponenti IP Telephony

R l Ti T t P t lReal-Time Transport Protocol

H.323 call signaling

Session Initiation Protocol (SIP)

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Voice over IP

IntroduzioneIntroduzioneIntroduzioneIntroduzioneEvoluzione delle reti e della telefonia verso le reti di nuova generazione.

Esigenze utenti ⇒ moltiplicazione e differenziazione dei servizi e dei possibili scenari di applicazione.

Domanda servizi ⇒ evolve verso reti di nuova generazione e t l f i f t t i ti l l lverso una telefonia fortemente innovativa con la quale la

competizione si sposterà dalle politiche di pricing alla diversificazione dei servizi.

In quest’ottica: la telefonia di nuova generazione (Next Generation Telephony – NGT ),

Evoluzione del servizio telefonico in cui il flusso voce diventa solo una parte di un servizio multimediale più complesso.

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Voice over IP

IntroduzioneIntroduzioneIntroduzioneIntroduzioneLa componente vocale e’ affiancata da flussi di informazioni di ptipo video e/o dati e l’accesso ai servizi può essere effettuato tramite terminali differenti e indipendentemente dalla postazione fisica dell’utente finalefisica dell utente finale.

Possibilità di personalizzazione del servizio da parte dell’utente finale, in particolare per quanto riguarda gli aspetti difinale, in particolare per quanto riguarda gli aspetti di sottoscrizione, configurazione, accesso ai servizi.

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Voice over IP

Once upon a time the PSTNOnce upon a time the PSTNOnce upon a time… the PSTNOnce upon a time… the PSTNCircuit switchingg

POTS, ISDN

Main characteristics:

SCP

circuit setupfixed bandwidthfixed delay

STP

SCPSS7

fixed delay

ISDNISDN

COCO BPXBPXPBXPBX COCOQ.931Q.931 Q.931Q.931

Codec Codec

Analog Analog

5Digital

Voice over IP

VoIP (initial) motivationsVoIP (initial) motivationsVoIP (initial) motivations VoIP (initial) motivations Billing

Initial stimulusIP is generally considered “free”; long distance phone calls were expensive

RegulationsLaws and authorities… deregulation

CostsVoIP Gateways (GWs) are cheaper than in the past Availability of very small GWs (i.e. 2 ports)

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Voice over IP

VoIP motivationsVoIP motivationsVoIP motivations VoIP motivations Integration (network/transport aspect)

the IP data traffic is dramatically growing with reference to voice trafficthe IP data traffic is dramatically growing with reference to voice trafficIn the backbones: the future highways are built for IP traffic!Fax services over IP (fax traffic today considered as telephone traffic will be likely replaced by email) For the ISPs: NAS with VoIP GW functions (Internet Telephony Service Provider - ITSP)

Integration (operation and management aspect)an integrated Voice/Data network can represent a large cost saving from the operational point of viewf l ithi t i th l ld h dl th D t dfor example within an enterprise the same people could handle the Data and Voice network… as the technology and the knowledge will be in common

ServicesServicesthe Voice/Data integration over IP networks opens the door to new advanced services

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Voice over IP

The main playersThe main playersThe traditional manufactures Data manufactures

The main playersThe main players

Ericsson, Lucent, Nortel, ...

Data manufacturesCisco, Ascend, 3Com, ...

New ComersNew ComersVocaltec, NetSpeak, E-Fusion, ViennaSys, ...

PSTN OperatorsAT&T, DT, Telekom Finland, ...

New ComersDelta3, ITXC, GRIC, OzMail, Qwest, USA

ISP and ITSPAOL, GEOnet, IDT ...

Talks, ...

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Voice over IP

Voice over IP e IP TelephonyVoice over IP e IP TelephonyVoice over IP e IP TelephonyVoice over IP e IP Telephony

VocalGatewayPSTN PSTN

RouterGSM

ISDN

GSMInternet/Intranet

IP networkISDN

I due termini sono spesso usati come sinonimi.. I due termini sono spesso usati come sinonimi.. pp

“Voice over IP”: supporto per comunicazioni voce utilizzando il protocollo su “Voice over IP”: supporto per comunicazioni voce utilizzando il protocollo su Internet/IntranetInternet/IntranetIl percorso telefonico puo’ essere interamente o parzialmente su IPIl percorso telefonico puo’ essere interamente o parzialmente su IPIl percorso telefonico puo essere interamente o parzialmente su IP.Il percorso telefonico puo essere interamente o parzialmente su IP.

“IP telephony” e “Internet telephony”: SERVIZIO“IP telephony” e “Internet telephony”: SERVIZIO

Almeno uno dei due terminali deve essere IP: il servizio non e’ PSTNAlmeno uno dei due terminali deve essere IP: il servizio non e’ PSTN

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Almeno uno dei due terminali deve essere IP: il servizio non e PSTNAlmeno uno dei due terminali deve essere IP: il servizio non e PSTN“IP telephony” non e’ solo Telephony over IP!“IP telephony” non e’ solo Telephony over IP!“IP telephony” include conferenza, video comunicazione, nuovi servizi“IP telephony” include conferenza, video comunicazione, nuovi servizi

Voice over IP

VoIP ScenariosVoIP ScenariosVoIP ScenariosVoIP Scenarios

phonephone--toto--phonephone

PSTN PSTNPSTNPSTN

IP networkPCPC--toto--phonephone

G tIP Network

Gateway Gateway

PCPC--toto--PCPC

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Voice over IP

Voice over IP: scenariVoice over IP: scenariVoice over IP: scenariVoice over IP: scenariIP-to-IP (PC-to-PC)

PC h d IP T l h ftPC scheda sonora + IP Telephony software (Internet Phone, Netmeeting, etc)Esempi: IP Telephony over Internet, corporate telephony serviceServizi opzionali: video, chat, condivisione

VoIP Terminal VoIP TerminalIP network

Servizi opzionali: video, chat, condivisione dati (data sharing)…

Gateway

IP-to-SCN/PSTN (PC-to-phone)Gateway per connettere la rete IP alla rete telefonica

IP networkPSTN

VoIP Terminaltelefonica

SCN-to-IP-to-SCN (phone-to-phone via an IP network))

Servono piu’ gatewaysLa rete IP puo’ essere o una intranet dedicata o InternetLe reti telefoniche complesse

PSTN

GatewayGateway

11IP network

yGateway

Switched Circuit Network (SCN)

Voice over IP

Structure of IPStructure of IPTelephonyTelephonyservicesservicesStructure of IPStructure of IPTelephonyTelephonyservicesservicesIP Telephony service provider (ITSP)

IP networks

ISP

PSTN

I t t t M d: Internet router or Telephony Server

: Modem

: Modem of ISP at access point

: Gateway of ITSP at access point: PSTN Switch

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y p

: Gateway towards PSTN

Voice over IP

VoIP/IP Telephony service basicsVoIP/IP Telephony service basicsVoIP/IP Telephony service basicsVoIP/IP Telephony service basicscontrol/signaling protocols- H.323 (H.225, H.245 + RAS)

ccontrol/signaling agents- H 323 Gatekeeper3 3 ( 5, 5 S)

- SIP- MGCP

- H.323 Gatekeeper- SIP Server- MGCP Call Agent

RTP flows

SIP/H.323 TerminalSIP/H.323 Terminal

gateway

PBX

SIP/H.323 Terminal

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PSTNPBX

Voice over IP

VoIP/IP Telephony modelVoIP/IP Telephony modelVoIP/IP Telephony modelVoIP/IP Telephony modelcontrol/signaling protocols- H.323 (H.225, H.245 + RAS)

control/signaling agents- H 323 Gatekeeper3 3 ( 5, 5 S)

- SIP- MGCP

- H.323 Gatekeeper- SIP Server- MGCP Call Agent

RTP flows

SIP/H.323 Terminal RTP flowsSIP/H.323 Terminal

gateway

PBX

SIP/H.323 Terminal

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PSTNPBX

Voice over IP

VoIP general architectureVoIP general architectureggSecurity, Addressing,AccountingVoice coding and

packetization

System Management

PSTN/IP Interworking

S hPSTN/IP Interworking:Speech

RepresentationAnd

TelephoneCall

Interworking: Gateways between networks

And Coding Control

Signaling ProtocolPacket TransportRequirements:

Latency, Packets loss

Signaling Protocol and Service Transparency regardless the

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Limited Delay, Jitterg

used technology

Voice over IP

GatewayGatewayGatewayGatewayIt is the “gateway” between an IP network and a Switched Circuit g yNetwork (SCN) (ex.: PBX, ISDN, POTS)

A Gateway takes a phone call from the SCN, digitizes it if not already digital, compresses and packetizes it into IP datagrams

Reverse for the other way (full-duplex)

It acts as gateway for both:user plane (data)

t l l ( i li )control plane (signaling)

Gateways are often based on PC server platforms

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Voice over IP

MarketsMarketsMarketsMarketsPC-to-PC over Internet

Corporate IP Telephony in place of PBX

Corporate toll-bypass

IP based public phone servicePC t hPC-to-phonephone-to-phone

New services!

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Voice over IP

Technical issuesTechnical issuesTechnical issuesTechnical issuesLarge Delayg y

Delay Jitter

Packet lengthPacket length

Lost of packets: replace lost packets by silence, extrapolate previous waveformprevious waveform

Echo cancellation

Silence suppressionSilence suppression

Address translation: Phone # to IP; directory servers

Telephony signaling: Different PBXs may use different signaling methods

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Voice over IP

Technical issues (Cont )Technical issues (Cont )Technical issues (Cont.)Technical issues (Cont.)Multiplexing: Subchannel multiplexing ==> Multiple voice calls in p g p g pone packet

Security: Firewalls may not allow incoming IP traffic

Insecurity of Internet

Voice compressionVoice compression

Charging. Not a technical issue! ;-)

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Voice over IP

EndEnd toto end delay budgetend delay budgetEndEnd--toto--end delay budgetend delay budget

AnnoyingPoor

Good150ms

Annoying

250ms 450ms

0

25 400 ms150dard

hold

s

ITU-T G.131 ITU-T G.114 ITU-T G.114

25 msEcho

Canceller

25 msEcho

Canceller

400 ms“Natural”

InteractionLimit

400 ms“Natural”

InteractionLimit

150 msDelay not Perceived

in Most Cases

150 msDelay not Perceived

in Most CasesStan

dTh

resh

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Voice over IP

EndEnd--toto--end delay componentsend delay componentsy py pEnd-to-end delay is given by :

Fixed component:• processing delay (voice coding and packetization) p g y ( g p )• propagation delay• serialization delay

Variable component• Delay introduced by the network (queuing delay, packet

processing)• Variable packet sizes• Variable packet sizes

21“Dejitter” buffers compensate the variable component

Voice over IP

RitardiRitardiRitardiRitardiProcessing delay: il tempo che il coder impiega per effettuare g y p p g pl’operazione di codifica. Questo tempo ovviamente dipende dal tipo di algoritmo di compressione utilizzato, dall’implementazione dello stesso e dalla potenza di calcolo della macchina Questodello stesso e dalla potenza di calcolo della macchina. Questo tipo di ritardo è in generale trascurabile rispetto agli altri due.

Frame size delay: per effettuare la codifica il coder ha bisogno diFrame size delay: per effettuare la codifica il coder ha bisogno di collezionare una serie di campioni in un buffer per poi elaborarli. Ovviamente questo buffer introduce un certo ritardo, detto frame size delay che dipende esclusivamente dal tipo di algoritmo disize delay, che dipende esclusivamente dal tipo di algoritmo di codifica utilizzato.

Look ahead delay: per migliorare la qualità della codifica alcuniLook ahead delay: per migliorare la qualità della codifica, alcuni algoritmi collezionano anche alcuni campioni vocali del frame successivo introducendo altro ritardo, il cosiddetto look ahead d l

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delay.

Voice over IP

The phone worksThe phone works why bother with VoIP?why bother with VoIP?The phone works The phone works —— why bother with VoIP?why bother with VoIP?User prospective Carrier prospectivep p

variable compression/quality

security through encryption

p p

silence suppression ==> low trafficsecurity through encryption

caller, talker identification

b tt i t f

integration of data and voice switching (only one network)

better user interface

low cost for international callsoperational advantages

integration of data and voice no local access fees

video, whiteboard... and new

gservices

new services!services!

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Voice over IP

Two Views of Internet TelephonyTwo Views of Internet TelephonyTwo Views of Internet TelephonyTwo Views of Internet TelephonyInternet telephony:Internet telephony:

primarily voice

look like phone system: ISDN signaling separate “stack”look like phone system: ISDN signaling, separate stack

interoperability with SS7

or SS7 migration to Internetor SS7 migration to Internet

Internet telephony:

VoIP: yet another Internet service

voice: small fraction of traffic in ten years

SS7: legacy, to be relegated to edges

integration with email, web

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multimedia

Voice over IP

IP Telephony basic componentsIP Telephony basic componentsIP Telephony basic componentsIP Telephony basic componentsCodec/decodec (audio, video)( , )

User planeRTP/RTCP/UDP/IPIP Multicast

Signaling: Call Setup, Tear-downH323 (ITU-T standard includes H.225.0, H.245, Q.931)Session Initiation Protocol (SIP) (IETF)MEGACOMEGACO

Gateways (H.323-to-PSTN, SIP-to-PSTN, etc.)

QoS: Resource ReservationQoS: Resource ReservationRSVPDifferntiated Services

25Policy issues: billing, firewall access, AAA (Authentication, Authorization, Accounting)

Voice over IP

VoIP protocol stackVoIP protocol stackVoIP protocol stackVoIP protocol stack

audio/video audio/video equipmentequipment session controlsession control

audio/video coding

RTP/RTCPH.323, SIP, others

IP ( + QoS/RSVP/Diffserv )

UDP TCP or UDP

layer 2 technologies (ATM, Ethernet, PPP, ...)

IP ( + QoS/RSVP/Diffserv )

* plus anything required for QoS guarantees (RSVP, Differentiated Services...)

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Voice over IP

ITUITU--T Study Group 16T Study Group 16y py pMultimedia Services and SystemsMultimedia Services and Systems

Responsible for studies relating to multimedia service definition p gand multimedia systems, including the associated terminals, modems, protocols and signal processing

ActivitiesH.323 (signaling protocol between terminals, gateways and gatekeepers)g p )H.225 and H.245 (Media Capabilities)H.248 (protocol between MG and MGC), in conjunction with IETF MEGACOMEGACOG.729 (8 kbit/s voice codec)

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Voice over IP

Main IETF Working Groups (WGs)Main IETF Working Groups (WGs)Main IETF Working Groups (WGs)Main IETF Working Groups (WGs)AVT - Audio/Video Transportp

formed to specify a protocol for real-time transmission of audio and video (RTP)the associated profile for audio/video conferences and payload formats (e.g. MPEG-4, PureVoice)

SIGTRAN - Signaling Transporttransport of packet-based PSTN signaling (such as Q.931 or SS7 ISUP) over IP Networks, between IP nodes such as a Signaling Gateway and Media Gateway Controller or Media Gateway

MEGACO - Media Gateway Control

MMUSIC - Multiparty Multimedia Session Controlto develop Internet standards track protocols to support Internet teleconferencing sessions

PINT PSTN d I t t I t t kiPINT - PSTN and Internet InternetworkingIPTEL - IP Telephony

to develop protocols for Internet telephony (ex. signaling and capabilities exchange) e.g. specification of the Call Processing syntax, specification of the service model, protocols, etc.

SIP - Session Initiation ProtocolSviluppo ed estensione di SIP

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Sviluppo ed estensione di SIP

Voice over IP

IETF Working Groups (WGs)IETF Working Groups (WGs)IETF Working Groups (WGs)IETF Working Groups (WGs)AVT - Audio/Video Transportp

Trasmissione Real-Time di segnali audio e video su IP

SIGTRAN - Signaling TransportInterlavoro/trasporto della segnalazione su IP

MEGACO - Media Gateway ControlVideoconferenze basate su IP (focus su annuncio, controllo...)

MMUSIC - Multiparty Multimedia Session ControlMMUSIC Multiparty Multimedia Session ControlVideoconferenze basate su IP (focus su annuncio, controllo...)

PINT - PSTN and Internet InternetworkingIntegrazione di servizi Internet e PSTN/IN (es: Click-to-Dial, ...)teg a o e d se te et e S / (es C c to a , )

IPTEL - IP TelephonyArchitettura e protocolli per Internet Telephony (H.323 o SIP)

SIP Session Initiation ProtocolSIP - Session Initiation ProtocolSviluppo ed estensione di SIP

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Voice over IP

IETF AVTIETF AVT Audio/Video TransportAudio/Video TransportIETF AVT IETF AVT -- Audio/Video TransportAudio/Video TransportThe Audio/Video Transport Working Group was formed to specify p g p p ya protocol for real-time transmission of audio and video over UDP and IP multicast

This is the Real-time Transport Protocol, RTP, together with its associated profile for audio/video conferences and payload format documentsdocuments

The payload formats currently under discussion include a number of media specific formats (MPEG-4, DTMF, PureVoice) and FEC p ( )techniques

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Voice over IP

IETF SIGTRANIETF SIGTRAN Signaling TransportSignaling TransportIETF SIGTRAN IETF SIGTRAN -- Signaling TransportSignaling TransportThe primary purpose of this working group is to address the t t f k t b d PSTN i li IP N t ktransport of packet-based PSTN signaling over IP Networks, taking into account functional and performance requirements of the PSTN signalingFor interworking with PSTN, IP networks will need to transport signaling such as Q.931 or SS7 ISUP messages between IP nodes such as a Signaling Gateway and Media Gateway Controller or Media GatewayController or Media Gateway

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Voice over IP

IETF MEGACOIETF MEGACO Media Gateway ControlMedia Gateway ControlIETF MEGACO IETF MEGACO -- Media Gateway ControlMedia Gateway Control

The working group will develop anThe working group will develop an architecture for controlling Media Gateways from external control

l t h M di

Signalling Signalling GatewayGateway

SCNSCN(es: SS7)(es: SS7) IP SigIP Sig

elements such as a Media Gateway Controller

A media gateway provides

yy

A media gateway provides conversion between the information carried on telephone i i d d k i d

Media Media GatewayGateway

ControllerControllercircuits and data packets carried over the Internet or over other IP networks

Co t o eCo t o e

Media Media GatewayGateway

RTP RTP StreamStream(es: PCM)(es: PCM)

Media Media StreamStream(es: PCM)(es: PCM)

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Voice over IP

IETF MMUSIC IETF MMUSIC -- Multiparty Multimedia Session Multiparty Multimedia Session p yp yControlControl

The Multiparty MUltimedia SessIon Control (MMUSIC) Working p y ( ) gGroup (WG) is chartered to develop Internet standards track protocols to support Internet teleconferencing sessions

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Voice over IP

IETF IPTELIETF IPTEL IP TelephonyIP TelephonyIETF IPTEL IETF IPTEL -- IP Telephony IP Telephony Before Internet telephony can become a widely deployed service, p y y p y ,a number of protocols must be deployed. These include signaling and capabilities exchange, but also include a number of "peripheral"protocols for providing related servicesperipheral protocols for providing related services.

The primary purpose of this working group is to develop protocols for Internet telephony (ex. signaling and capabilities exchange)for Internet telephony (ex. signaling and capabilities exchange)

Among various tasks, there is:specification of the Call Processing syntax,p g y ,specification of the service model,specification of the Gateway Attribute Distribution Protocol (attributes such as PSTN connectivity supported codecs etc )(attributes such as PSTN connectivity, supported codecs, etc.)

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SIPSIPSS

Voice over IP

IETF SIPIETF SIP Session Initiation ProtocolSession Initiation ProtocolIETF SIP IETF SIP -- Session Initiation ProtocolSession Initiation ProtocolThe Session Initiation Protocol (SIP) working group is chartered to ( ) g g pdevelop SIP protocol, currently specified as proposed standard RFC 2543

SIP is a text-based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users

S h i i l d i id h t i t ti dSuch sessions include voice, video, chat, interactive games, and virtual reality.

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Voice over IP

SIP Session Initation ProtocolSIP Session Initation ProtocolSIP Session Initation ProtocolSIP Session Initation ProtocolProtocollo di segnalazione e di controllo a livello applicativo che permette di stabilire modificare e terminare sessioni multimedialipermette di stabilire modificare e terminare sessioni multimediali (conferenze, videotelefonia su Internet…) o semplici chiamate.IETF (Intenet Engineering Task Force) nel ’99 come RFC 2543; in continuo aggiornamento (RFC 2543 9bis)continuo aggiornamento, (RFC 2543 9bis).Protocollo Client/Server:

Terminali SIP (end-point intelligenti che supportano comunicazioni real-time in grado di elaborare le informazioni riguardanti la sessione a cuitime, in grado di elaborare le informazioni riguardanti la sessione a cui desiderano partecipare; Proxy Server (intermediario tra terminale chiamante e chiamato, agisce sia da client che da server e si occupa della distribuzione dei messaggi SIP).

IntroduzioneNuovi messaggi (es. INVITE)SIP URL. Gli indirizzi SIP, per identificare un utente o un servizio, sono del formato user@host dove user può essere il nome dell’utente o il suo numformato user@host, dove user può essere il nome dell utente o il suo num. tel. e host il nome di dominio o l’indirizzo di rete.

SIP e’ standard in UMTS, (anche in Microsoft .net), Allo studio nuove estensioni per supporto di servizi 3G

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Allo studio nuove estensioni per supporto di servizi 3G.

Voice over IP

IL PROTOCOLLO SIPIL PROTOCOLLO SIPIL PROTOCOLLO SIPIL PROTOCOLLO SIP

ProxyServer

Protocollo di segnalazione per sessioni multimediali su IP

Proxy Server

Definito in ambito IETF

Definisce nuovi componenti

Internet

rRouterRoute

r Introduce SIP URL: user@host(es.0362-12345@pippo.com)

TerminaliSIP

Location

Adottato come standard nel mondo UMTS

Server RedirectServer

stanno nascendo nuove estensioni

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del protocollo per il supporto di nuovi servizi

Voice over IP

Analizzatore di protocolloAnalizzatore di protocolloAnalizzatore di protocolloAnalizzatore di protocolloEthereal

Senza estensione H.323: http://www.ethereal.com/

Estensione H 323: http://www voice2sniff org/Estensione H.323: http://www.voice2sniff.org/

Passi per l’installazionehttp://www.ethereal.com/distribution/win32/ e scaricare i fileshttp://www.ethereal.com/distribution/win32/ e scaricare i files eseguibilihttp://winpcap.polito.it/install/default.htm e scaricare WinPcap 2.3 (se avete WinXP)(se avete WinXP)Installare WinPcapInstallare Etherealhttp://www.voice2sniff.org i pluging H323 per Ethereal e seguire le procedure di installazione come indicato nel file Readme

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Voice over IP

SoftwareSoftwareSoftwareSoftwareSistema VOCAL di VOVIDA

http://www.vovida.org/

Altri possono essere trovati al sitoAltri possono essere trovati al sito

http://www.iptelephony.org/GIP/vendors/client-phones/

Ubi it SIP Cli tUbiquity SIP Client

http://www.ubiquity.net/useragent.php

Password va richiesta a Ubiquity (è Free)

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