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VOIP, VOATM, VOFR
1. VOIP
 Voice over Internet Protocol, tiež nazývané
VoIP, IP Telefónia, Internetová telefónia, je
prenos komunikácie uskutočňovanej ľudským
hlasom cez Internet alebo inú sieť založenú na
protokole IP.
 Protokoly používané na prenos hlasových
signálov cez IP sieť = VoIP protokoly
 Základ - Network Voice Protocol (1973)
navrhnutého pre sieť ARPANET
1.1 Funkcia
• Prichádzajúce telefónne hovory môžu byť automaticky smerované na VoIP
telefón, nezávisle na tom, kde sa nachádzate.
•Vo viacerých krajinách (USA, Veľká Británia, atď.) sú k dispozícii bezplatne
použiteľné telefónne čísla pre použitie vo VoIP.
•Pracovníci call centier môžu pri použití VoIP pracovať z ľubovoľného miesta,
kde je k dispozícii dostatočne stabilné internetové pripojenie.
•Mnohé VoIP balíky služieb obsahujú funkcie verejných sietí, ktoré sú bežne
spoplatňované osobitne, prípadne sú miestnym operátorom osobitne
spoplatňované, ako napríklad konferenčné hovory, presmerovanie hovoru,
automatické opakovanie vytáčania a pod.
•VoIP telefóny dokážu spájať viacero služieb dostupných cez Internet vrátane
videokonferencií, prenosu dát popri hovore, správy telefónnych a adresových
zoznamov a oznamovania online dostupnosti zvolených komunikačných
partnerov.
1.2 VOIP zapojenie
ATA – analógový terminálový adaptér
1.3 Technické detaily
Dva najhlavnejšie súperiace štandardy pre VoIP sú Session
Initiation Protocol (SIP), vyvinutý pod hlavičkou organizácie
IETF, a štandard ITU s označením H.323. Na počiatku bol
populárnejším H.323, čo je štandard vychádzajúci z
telekomunikačného prostredia, v súčasnosti je už v popredí
SIP, s ktorým sa ráta už aj v ústredniach typu IMS.
1.5 SIP
The SIP (Session Initiation Protocol) is a text-based protocol, similar to the HTTP and SMTP,
designed for initiating, maintaining and terminating of interactive communication sessions
between users. Such sessions include voice, video, chat, interactive games, and virtual
reality.
The SIP defines and uses the following components:
• UAC (User agent client) – client in the terminal that initiates SIP signalling
• UAS (User agent server) – server in the terminal that responds to the SIP signalling from the
UAC
• UA (User Agent) – SIP network terminal (SIP telephones, or gateway to other networks),
contains UAC and UAS
• Proxy server – receives connection requests from the UA and transfers them to another
proxy server if the particular station is not in its administration
• Redirect server – receives connection requests and sends them back to the requester
including destination data instead of sending them to the calling party
• Location Server – receives registration requests from the UA and updates the terminal
database with them.
All server sections (Proxy, Redirect, Location) are typically available on a single physical
machine called proxy server, which is responsible for client database maintenance,
connection establishing, maintenance and termination, and call directing.
Basic messages sent in the SIP environment:
•INVITE – connection establishing request
•ACK – acknowledgement of INVITE by the final message receiver
•BYE – connection termination
•CANCEL – termination of non-established connection
•REGISTER – UA registration in SIP proxy
•OPTIONS – inquiry of server options
Answers to SIP messages are in the digital format like in the http protocol. Here are the most
important ones:
1XX – information messages (100 – trying, 180 – ringing, 183 – progress)
2XX – successful request completion (200 – OK)
3XX – call forwarding, the inquiry should be directed elsewhere (302 – temporarily moved,
305 – use proxy)
4XX – error (403 – forbidden)
5XX – server error (500 – Server Internal Error, 501 – not implemented)
6XX – global failure (606 – Not Acceptable)
1.6 Porovnanie H.323 vs. SIP
H.323
SIP
• vytvorený pre mediálnu
komunikáciu (videokonferencie a
pod), robustný
• vytvára relácie medzi dvoma
bodmi, nepodporuje
multimediálne konferencie
• dokáže reagovať na chyby
sieťových zariadení
• neodstraňuje poruchy sieť. zar.
• na transport dát využíva
RTP/RTCP, SRTP
• na vytvorenie spojenia využíva
UDP
• podporuje všetky kodeky,
štandardizované alebo
proprietárne
•na transport dát využíva
RTP/RTCP, SRTP
• na vytvorenie spojenia využíva
UDP
• podporuje kodeky registrované
v IANA
STUN
STUN
 Simple Traversal of UDP through NATs
(STUN), is a network protocol allowing a client
behind a NAT (Network Address Translator) to
find out its public address, the type of NAT it is
behind and the internet-side port associated by
the NAT with a particular local port. This
information is used to set up UDP (User
Datagram Protocol) communication between two
hosts that are both behind NAT routers. The
protocol is defined in RFC 3489.
IAX - Inter-Asterisk eXchange
 IAX2 is a VoIP protocol that usually carries
both signalling and data on the same path.
The commands and parameters are sent
binary and any extension has to have a
new numeric code allocated. Historically
this was modeled after the internal data
passing of Asterisk modules
IAX2
 IAX2 uses a single UDP data stream (usually on port 4569)
to communicate between endpoints, both for signaling and
data. The voice traffic is transmitted in-band, making IAX2
easier to firewall and more likely to work behind network
address translation. This is in contrast to SIP, H.323 and
Media Gateway Control Protocol which are using an out-ofband RTP stream to deliver information.
 IAX2 supports trunking, multiplexing channels over a single
link. When trunking, data from multiple calls are merged into
a single set of packets, meaning that one IP datagram can
deliver information for more than one call, reducing the
effective IP overhead without creating additional latency.
This is a big advantage for VoIP users, where IP headers
are large percentage of the bandwidth usage.
SIP vs IAX
 - Bandwidth
The bandwidth uses by IAX is less than the one uses by SIP
since the messages are binary instead of text messages (SIP).
IAX also tries to reduce the headers of the messages reducing
therefore the bandwidth used.
- NAT
Signaling and data travel togheter in IAX avoiding the problems
of NAT that usually appear in SIP. Signaling and data in SIP travel
using different protocols and that is why NAT problems appears.
Audio stream have to pass through routers and firewalls. SIP usually
needs a STUN server to avoid these problems.
- Standarization and use
SIP is a protocol standardized by the IETF long time ago and it is
widely used by the equipment and software manufacturers. IAX is
still being standardized and for that reason not many devices
can use it nowadays
SIP vs IAX
 - Ports used
IAX uses only one port (4569) to send signalling and data of all the calls.
To do it IAX use a trunking system. IAX multiplexes signaling and multiple
media streams over a single User Datagram Protocol (UDP). SIP, otherwise,
uses one port (5060) for signalling and 2 RTP ports for each audio
connection (at least 3 ports). For example, if we have 100 simultaneous calls
we should use 200 RTP ports and one port for signalling (5060) . IAX uses
only one port for everything (4569)
- Audio flow when using a server
If SIP is using a server signaling messages always pass through the server
but audio messages (RTP flow) can travel end to end without passing
through the server. In IAX, signaling and data must pass always through
IAX server. This increases the bandwidth need by the IAX servers when there
are many simultaneous calls.
- Other functionalities
IAX is a protocol developed to VoIP and video transmission and it has
interesting functionalities, for example, the possibility to send or receive
dialplans. These funtionalities are very interesting if using Asterisk PBX.
SIP is a general porpouse protocol and can transmit any information
and not only audio or video.
2. VOATM
•Prenáša hlas po ATM sieti
• Fragmentácia dát na základe
ATM buniek
• na zníženie zdržania pri
prenose použité DBCES
(Dynamic Bandwith Circuit
Emulation Service)
•
3. VOFR
 Voice Over Frame relay
 Fragmentácia – dátové pakety sa delia na
menšie časti, ktoré sa rýchlejšie posielajú
po sieti
 dáta prechádzajú cez VFRAD (Voice
Frame Relay Access Devices ) – podľa
priority smeruje pakety
Protokolové štandardy:
• ADPCM (Adaptive Differential Pulse Code Modulation) – kóduje analógový
signál na binárny
• CS-ACELP (Conjugate Structure-Algebraic Code Excited Linear Prediction) kóduje analógový signál na binárny
Použitý zdroj informácií
• cisco.netacad.net
• www.earchiv.cz/
• wikipedia.org