Voice over IP

Voice over IP
Roadmap
z Introduzione
z IP Telephony/VoIP
z Componenti IP Telephony
Voice over IP
z Real-Time Transport Protocol
z H.323 call signaling
z Session Initiation Protocol (SIP)
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Voice over IP
Introduzione
Voice over IP
Introduzione
z Evoluzione delle reti e della telefonia verso le reti di nuova
generazione.
z La componente vocale e’ affiancata da flussi di informazioni di
tipo video e/o dati e l’accesso ai servizi può essere effettuato
tramite terminali differenti e indipendentemente dalla postazione
fisica dell’utente finale.
z Esigenze utenti ⇒ moltiplicazione e differenziazione dei servizi e
dei possibili scenari di applicazione.
z Possibilità di personalizzazione del servizio da parte dell’utente
finale, in particolare per quanto riguarda gli aspetti di
sottoscrizione, configurazione, accesso ai servizi.
z Domanda servizi ⇒ evolve verso reti di nuova generazione e
verso una telefonia fortemente innovativa con la quale la
competizione si sposterà dalle politiche di pricing alla
diversificazione dei servizi.
z In quest’ottica: la telefonia di nuova generazione (Next
Generation Telephony – NGT ),
¾ 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
Once upon a time…
time… the PSTN
Voice over IP
VoIP (initial) motivations
Billing
z Circuit switching
¾ Initial stimulus
¾ IP is generally considered “free”; long distance phone calls were expensive
¾ POTS, ISDN
z Main characteristics:
¾ circuit setup
¾ fixed bandwidth
¾ fixed delay
Regulations
¾ Laws and authorities… deregulation
SCP
SS7
STP
STP
Costs
PBX
PBX
Q.931
CO
CO
CO
CO
ISDN
Analog
Q.931
BPX
BPX
¾ VoIP Gateways (GWs) are cheaper than in the past
¾ Availability of very small GWs (i.e. 2 ports)
ISDN
Analog
Codec
Codec
Digital
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Voice over IP
VoIP motivations
Voice over IP
The main players
The traditional
manufactures
Integration (network/transport aspect)
¾ the IP data traffic is dramatically growing with reference to voice traffic
¾ In the backbones: the future highways are built for IP traffic!
¾ Fax services over IP (fax traffic today considered as telephone traffic will be
likely replaced by email)
¾ For the ISPs: NAS with VoIP GW functions (Internet Telephony Service Provider
- ITSP)
Ericsson, Lucent, Nortel,
...
Data manufactures
Cisco, Ascend, 3Com, ...
New Comers
Vocaltec, NetSpeak, E-Fusion, ViennaSys, ...
Integration (operation and management aspect)
PSTN Operators
¾ an integrated Voice/Data network can represent a large cost saving from the
operational point of view
¾ for example within an enterprise the same people could handle the Data and
Voice network… as the technology and the knowledge will be in common
AT&T, DT, Telekom Finland, ...
New Comers
Services
ISP and ITSP
¾ the Voice/Data integration over IP networks opens the door to new advanced
services
Delta3, ITXC, GRIC, OzMail, Qwest, USA
Talks, ...
AOL, GEOnet, IDT ...
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Voice over IP
Voice over IP
Voice over IP e IP Telephony
PSTN
GSM
VoIP Scenarios
Vocal
Gateway
phonephone-toto-phone
PSTN
Router
Internet/Intranet
PSTN
PSTN
PSTN
PSTN
GSM
IP network
ISDN
ISDN
PCPC-toto-phone
IP
IPnetwork
Network
z I due termini sono spesso usati come sinonimi..
sinonimi..
Gateway
Gateway
z “Voice over IP”
IP”: supporto per comunicazioni voce utilizzando il protocollo su
Internet/Intranet
Il percorso telefonico puo’
puo’ essere interamente o parzialmente su IP.
z “IP telephony”
telephony” e “Internet telephony”
telephony”: SERVIZIO
PCPC-toto-PC
z Almeno uno dei due terminali deve essere IP: il servizio non e’
e’ PSTN
¾ “IP telephony”
telephony” non e’
e’ solo Telephony over IP!
¾ “IP telephony”
telephony” include conferenza,
conferenza, video comunicazione,
comunicazione, nuovi servizi
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Voice over IP
Voice over IP: scenari
Structure of IP Telephony services
IP Telephony service provider (ITSP)
z IP-to-IP (PC-to-PC)
¾ PC scheda sonora + IP Telephony software
(Internet Phone, Netmeeting, etc)
¾ Esempi: IP Telephony over Internet,
corporate telephony service
¾ Servizi opzionali: video, chat, condivisione
dati (data sharing)…
Voice over IP
VoIP Terminal
IP network
VoIP Terminal
IP networks
ISP
Gateway
z IP-to-SCN/PSTN (PC-to-phone)
PSTN
IP network
¾ Gateway per connettere la rete IP alla rete
telefonica
z SCN-to-IP-to-SCN (phone-to-phone
via an IP network)
¾ Servono piu’ gateways
¾ La rete IP puo’ essere o una intranet
dedicata o Internet
¾ Le reti telefoniche complesse
PSTN
VoIP Terminal
: Internet router or
Telephony Server
PSTN
: PSTN Switch
Gateway
Gateway
: Modem of ISP at access point
: Gateway of ITSP at access point
: Gateway towards PSTN
IP network
Switched Circuit Network (SCN)
: Modem
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Voice over IP
Voice over IP
VoIP/IP
VoIP/IP Telephony service basics
control/signaling
control/signaling protocols
protocols
-- H.323
H.323 (H.225,
(H.225, H.245
H.245 ++ RAS)
RAS)
-- SIP
SIP
-- MGCP
MGCP
VoIP/IP
VoIP/IP Telephony model
control/signaling
control/signaling protocols
protocols
-- H.323
H.323 (H.225,
(H.225, H.245
H.245 ++ RAS)
RAS)
-- SIP
SIP
-- MGCP
MGCP
ccontrol/signaling
ccontrol/signaling agents
agents
-- H.323
H.323 Gatekeeper
Gatekeeper
-- SIP
SIP Server
Server
-- MGCP
MGCP Call
Call Agent
Agent
control/signaling
control/signaling agents
agents
-- H.323
H.323 Gatekeeper
Gatekeeper
-- SIP
SIP Server
Server
-- MGCP
MGCP Call
Call Agent
Agent
RTP flows
RTP flows
SIP/H.323 Terminal
SIP/H.323 Terminal
RTP flows
SIP/H.323 Terminal
SIP/H.323 Terminal
gateway
gateway
PBX
PBX
PSTN
PSTN
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Voice over IP
Voice over IP
VoIP general architecture
Security, Addressing,
Accounting
Voice coding and
packetization
System Management
PSTN/IP Interworking
Speech
Representation
And
Coding
Requirements:
Latency,
Packets loss
Limited Delay, Jitter
Telephone
Call
Control
Packet Transport
Gateway
z It is the “gateway” between an IP network and a Switched Circuit
Network (SCN) (ex.: PBX, ISDN, POTS)
z A Gateway takes a phone call from the SCN, digitizes it if not
already digital, compresses and packetizes it into IP datagrams
PSTN/IP
Interworking:
Gateways
between networks
z Reverse for the other way (full-duplex)
z It acts as gateway for both:
¾ user plane (data)
¾ control plane (signaling)
z Gateways are often based on PC server platforms
Signaling Protocol
and Service
Transparency
regardless the
used technology
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Voice over IP
Voice over IP
Markets
Technical issues
z PC-to-PC over Internet
z Large Delay
z Delay Jitter
z Corporate IP Telephony in place of PBX
z Packet length
z Lost of packets: replace lost packets by silence, extrapolate
previous waveform
z Corporate toll-bypass
z Echo cancellation
z IP based public phone service
z Silence suppression
¾ PC-to-phone
¾ phone-to-phone
z Address translation: Phone # to IP; directory servers
z Telephony signaling: Different PBXs may use different signaling
methods
z New services!
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Voice over IP
Voice over IP
Technical issues (Cont.)
EndEnd-toto-end delay budget
z Multiplexing: Subchannel multiplexing ==> Multiple voice calls in
one packet
Poor
z Security: Firewalls may not allow incoming IP traffic
Annoying
Good
z Insecurity of Internet
150ms
z Voice compression
250ms
450ms
0
Standard
Thresholds
z Charging. Not a technical issue! ;-)
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ITU-T G.131
25
25ms
ms
Echo
Echo
Canceller
Canceller
ITU-T G.114
150
150ms
ms
Delay
Delaynot
notPerceived
Perceived
in
inMost
MostCases
Cases
ITU-T G.114
400
400ms
ms
“Natural”
“Natural”
Interaction
Interaction
Limit
Limit
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Voice over IP
Voice over IP
EndEnd-toto-end delay components
Ritardi
z End-to-end delay is given by :
z Processing delay: il tempo che il coder impiega per effettuare
l’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. Questo
tipo di ritardo è in generale trascurabile rispetto agli altri due.
¾ Fixed component:
• processing delay (voice coding and packetization)
• propagation delay
• serialization delay
z Frame 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 di
codifica utilizzato.
¾ Variable component
• Delay introduced by the network (queuing delay, packet
processing)
• Variable packet sizes
z Look 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
delay.
“Dejitter” buffers compensate the variable component
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Voice over IP
Voice over IP
The phone works — why bother with VoIP?
VoIP?
Two Views of Internet Telephony
User prospective
Carrier prospective
Internet
z variable compression/quality
z silence suppression ==> low
traffic
z primarily voice
z security through encryption
z caller, talker identification
z better user interface
z low cost for international calls
z no local access fees
z video, whiteboard... and new
services!
telephony:
z look like phone system: ISDN signaling, separate “stack”
z integration of data and voice
switching (only one network)
z interoperability with SS7
z or SS7 migration to Internet
z operational advantages
Internet telephony:
z integration of data and voice
services
z VoIP: yet another Internet service
z voice: small fraction of traffic in ten years
z new services!
z SS7: legacy, to be relegated to edges
z integration with email, web
z multimedia
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Voice over IP
IP Telephony basic components
Voice over IP
VoIP protocol stack
z Codec/decodec (audio, video)
z User plane
audio/video
equipment
¾ RTP/RTCP/UDP/IP
¾ IP Multicast
session control
audio/video coding
H.323, SIP, others
z Signaling: Call Setup, Tear-down
RTP/RTCP
¾ H323 (ITU-T standard includes H.225.0, H.245, Q.931)
¾ Session Initiation Protocol (SIP) (IETF)
¾ MEGACO
UDP
TCP or UDP
IP ( + QoS/RSVP/Diffserv )
z Gateways (H.323-to-PSTN, SIP-to-PSTN, etc.)
layer 2 technologies (ATM, Ethernet, PPP, ...)
z QoS: Resource Reservation
¾ RSVP
¾ Differntiated Services
* plus anything required for QoS guarantees (RSVP, Differentiated Services...)
z Policy issues: billing, firewall access, AAA (Authentication,
Authorization, Accounting)
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Voice over IP
Voice over IP
ITUITU-T Study Group 16
Multimedia Services and Systems
Main IETF Working Groups (WGs
(WGs))
z AVT - Audio/Video Transport
z Responsible for studies relating to multimedia service definition
and multimedia systems, including the associated terminals,
modems, protocols and signal processing
¾ formed to specify a protocol for real-time transmission of audio and video (RTP)
¾ the associated profile for audio/video conferences and payload formats (e.g. MPEG-4, PureVoice)
z SIGTRAN - Signaling Transport
z Activities
¾ transport 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
z MEGACO - Media Gateway Control
¾ H.323 (signaling protocol between terminals, gateways and
gatekeepers)
¾ H.225 and H.245 (Media Capabilities)
¾ H.248 (protocol between MG and MGC), in conjunction with IETF
MEGACO
¾ G.729 (8 kbit/s voice codec)
z MMUSIC - Multiparty Multimedia Session Control
¾ to develop Internet standards track protocols to support Internet teleconferencing sessions
z PINT - PSTN and Internet Internetworking
z IPTEL - IP Telephony
¾ to develop protocols for Internet telephony (ex. signaling and capabilities exchange)
¾ e.g. specification of the Call Processing syntax, specification of the service model, protocols, etc.
z SIP - Session Initiation Protocol
Sviluppo ed estensione di SIP
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Voice over IP
IETF Working Groups (WGs
(WGs))
Voice over IP
IETF AVT - Audio/Video Transport
z AVT - Audio/Video Transport
z The Audio/Video Transport Working Group was formed to specify
a protocol for real-time transmission of audio and video over UDP
and IP multicast
Trasmissione Real-Time di segnali audio e video su IP
z SIGTRAN - Signaling Transport
Interlavoro/trasporto della segnalazione su IP
z This is the Real-time Transport Protocol, RTP, together with its
associated profile for audio/video conferences and payload format
documents
z MEGACO - Media Gateway Control
Videoconferenze basate su IP (focus su annuncio, controllo...)
z MMUSIC - Multiparty Multimedia Session Control
Videoconferenze basate su IP (focus su annuncio, controllo...)
z The payload formats currently under discussion include a number
of media specific formats (MPEG-4, DTMF, PureVoice) and FEC
techniques
z PINT - PSTN and Internet Internetworking
Integrazione di servizi Internet e PSTN/IN (es: Click-to-Dial, ...)
z IPTEL - IP Telephony
Architettura e protocolli per Internet Telephony (H.323 o SIP)
z SIP - Session Initiation Protocol
Sviluppo ed estensione di SIP
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Voice over IP
Voice over IP
IETF SIGTRAN - Signaling Transport
IETF MEGACO - Media Gateway Control
z The primary purpose of this working group is to address the
transport of packet-based PSTN signaling over IP Networks,
taking into account functional and performance requirements of
the PSTN signaling
zThe working group will develop an
architecture for controlling Media
Gateways from external control
elements such as a Media
Gateway Controller
z For 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 Gateway
SCN
(es:
es: SS7)
zA media gateway provides
conversion between the
information carried on telephone
circuits and data packets carried
over the Internet or over other IP
networks
IP Sig
Media
Media
Gateway
Gateway
Controller
Controller
Media
Stream
(es:
es: PCM)
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Signalling
Signalling
Gateway
Gateway
Media
Media
Gateway
Gateway
RTP
Stream
(es:
es: PCM)
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Voice over IP
IETF MMUSIC - Multiparty Multimedia Session
Control
z The Multiparty MUltimedia SessIon Control (MMUSIC) Working
Group (WG) is chartered to develop Internet standards track
protocols to support Internet teleconferencing sessions
Voice over IP
IETF IPTEL - IP Telephony
z Before Internet telephony can become a widely deployed service,
a number of protocols must be deployed. These include signaling
and capabilities exchange, but also include a number of
"peripheral"protocols for providing related services.
z The primary purpose of this working group is to develop protocols
for Internet telephony (ex. signaling and capabilities exchange)
z Among various tasks, there is:
¾ specification of the Call Processing syntax,
¾ specification of the service model,
¾ specification of the Gateway Attribute Distribution Protocol
(attributes such as PSTN connectivity, supported codecs, etc.)
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Voice over IP
IETF SIP - Session Initiation Protocol
z The Session Initiation Protocol (SIP) working group is chartered to
develop SIP protocol, currently specified as proposed standard
RFC 2543
SIP
z SIP is a text-based protocol, similar to HTTP and SMTP, for
initiating interactive communication sessions between users
z Such sessions include voice, video, chat, interactive games, and
virtual reality.
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Voice over IP
Voice over IP
IL PROTOCOLLO SIP
SIP Session Initation Protocol
z Protocollo di segnalazione e di controllo a livello applicativo che
permette di stabilire modificare e terminare sessioni multimediali
(conferenze, videotelefonia su Internet…) o semplici chiamate.
z Protocollo di segnalazione per
sessioni multimediali su IP
Proxy
Server
z IETF (Intenet Engineering Task Force) nel ’99 come RFC 2543; in
continuo aggiornamento, (RFC 2543 9bis).
z Definito in ambito IETF
z Protocollo Client/Server:
¾ Terminali SIP (end-point intelligenti che supportano comunicazioni realtime, in grado di elaborare le informazioni riguardanti la sessione a cui
desiderano partecipare;
¾ Proxy Server (intermediario tra terminale chiamante e chiamato, agisce sia
da client che da server e si occupa della distribuzione dei messaggi SIP).
Proxy
Serve
r
Route
r
Router
z Definisce nuovi componenti
z Introduce SIP URL: user@host
Internet
([email protected])
z Introduzione
Terminali
SIP
¾ Nuovi messaggi (es. INVITE)
¾ 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 num.
tel. e host il nome di dominio o l’indirizzo di rete.
Locatio
n
Server
z SIP e’ standard in UMTS, (anche in Microsoft .net),
z Adottato come standard nel mondo
UMTS
Redirect
Server
stanno nascendo nuove estensioni
del protocollo per il supporto di nuovi
servizi
z Allo studio nuove estensioni per supporto di servizi 3G.
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Voice over IP
Voice over IP
Analizzatore di protocollo
Software
z Ethereal
z Sistema VOCAL di VOVIDA
z
Senza estensione H.323: http://www.ethereal.com/
z
http://www.vovida.org/
z
Estensione H.323: http://www.voice2sniff.org/
z
Altri possono essere trovati al sito
Passi per l’installazione
z
http://www.iptelephony.org/GIP/vendors/client-phones/
¾ http://www.ethereal.com/distribution/win32/ e scaricare i files
eseguibili
¾ http://winpcap.polito.it/install/default.htm e scaricare WinPcap 2.3
(se avete WinXP)
¾ Installare WinPcap
¾ Installare Ethereal
¾ http://www.voice2sniff.org i pluging H323 per Ethereal e seguire le
procedure di installazione come indicato nel file Readme
z Ubiquity SIP Client
z
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z
http://www.ubiquity.net/useragent.php
z
Password va richiesta a Ubiquity (è Free)
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