1. VOIP, VOATM, VOFR

2. 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. ľudským hlasomInternetsieťIPľudským hlasomInternetsieťIP  Protokoly používané na prenos hlasových signálov cez IP sieť = VoIP protokoly Protokoly signálov Protokoly signálov  Základ - Network Voice Protocol (1973) navrhnutého pre sieť ARPANET Network Voice Protocol1973ARPANETNetwork Voice Protocol1973ARPANET

3. 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.USAVeľká Británia Pracovníci call centier môžu pri použití VoIP pracovať z ľubovoľného miesta, kde je k dispozícii dostatočne stabilné internetové pripojenie.call centier 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.online 1.1 Funkcia

4. 1.2 VOIP zapojenie ATA – analógový terminálový adaptér

5. 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.Session Initiation Protocol IETFITUH.323

24. 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.

25. 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)

27. 1.6 Porovnanie H.323 vs. SIP H.323 vytvorený pre mediálnu komunikáciu (videokonferencie a pod), robustný dokáže reagovať na chyby sieťových zariadení na transport dát využíva RTP/RTCP, SRTP na vytvorenie spojenia využíva UDP podporuje všetky kodeky, štandardizované alebo proprietárne SIP vytvára relácie medzi dvoma bodmi, nepodporuje multimediálne konferencie neodstraňuje poruchy sieť. zar. na transport dát využíva RTP/RTCP, SRTP na vytvorenie spojenia využíva UDP podporuje kodeky registrované v IANA

28. STUN

29. 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. network protocolNATUDPRFC 3489network protocolNATUDPRFC 3489

30. 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 VoIPAsteriskVoIPAsterisk

31. 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-of- band RTP stream to deliver information. UDPsignalingin-bandfirewallnetwork address translationSIPH.323 Media Gateway Control Protocolout-of- bandRTPUDPsignalingin-bandfirewallnetwork address translationSIPH.323 Media Gateway Control Protocolout-of- bandRTP  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. trunkingmultiplexingtrunkingpacketsIPVoIPbandwidthtrunkingmultiplexingtrunkingpacketsIPVoIPbandwidth

32. 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

33. 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.

34. 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)

41. 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

43. 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

44. Použitý zdroj informácií cisco.netacad.net cisco.netacad.net www.earchiv.cz/ www.earchiv.cz/ www.earchiv.cz/ wikipedia.org wikipedia.org