Internetes m

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Internetes médiakommunikációVoIP

• Takács György
• 10. eloadás
• 2009. 05. 06.

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What is VoIP?

• VOIP is an acronym for Voice Over Internet
  Protocol, or in more common terms phone service
  over the Internet. If you have a reasonable
  quality Internet connection you can get phone
  service delivered through your Internet
  connection instead of from your local phone
  company. Some people use VOIP in addition to
  their traditional phone service, since VOIP
  service providers usually offer lower rates than
  traditional phone companies, but sometimes
  doesn't offer phone directory listings, or other
  common phone services. While many VoIP providers
  offer these services, consistent industry-wide
  means of offering these are still developing.

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What is VoIP?

• VoIP SKYPE

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What is VoIP?

• Hard Phones
• Cordless Hard Phones
• Dialup Hard Phones
• WLAN or WiFi Phones
• Hard Phones (voice and video)
• Soft Phones (voice only)
• Soft Phones (voice and video)

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Hard Phones

• Call Control supports 3Com NBX platforms
• Power over Ethernet IEEE 802.3af support
• Network Connectivity 10/100 switched Ethernet
  port
• Codecs G.711, ADPCM, G729a/b (requires system
  software support
• QoS 802.1p, IP-ToS, and VLAN support
• Jitter Buffer Adaptive
• DHCP Supports option 184
• RTP Frame Size 20/30 ms
• Silence Suppression Supported with G.729b
  codec

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Cordless Hard Phones

• Cordless phones (e.g.DECT) with IP interface on
  their base station.

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Dialup Hard Phones

• A dialup hard phone is a hard phone with a
  built-in modem instead of the Ethernet port. It
  will connect through the modem via a dialup
  internet service to a remote VoIP server and is
  therefore self contained. It does not require a
  personal computer nor any software to be run on a
  personal computer to make and receive VoIP phone
  calls. All that is required is a phone line and a
  dialup internet account. Dialup hard phones are
  popular in countries where there is very little
  broadband infrastructure yet.

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WLAN or WiFi Phones

• A WLAN or WiFi phone is a hard phone with a
  built-in WiFi transceiver unit instead of an
  Ethernet port to connect to a WiFi base station
  and from there to a remote VoIP server. It does
  not require a personal computer nor any software
  to be run on a personal computer to make and
  receive VoIP phone calls. All that is required is
  access to a WiFi base station.

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Hard Phones (voice and video)

• Hard phones with video telephony support.

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Soft Phones (voice only)

• A soft phone is an IP telephone in software. It
  can be installed on a personal computer and
  function as an IP phone. Soft phones require
  appropriate audio hardware to be present on the
  personal computer they run. This can either be a
  sound card with speakers or earphones and a
  microphone, or, alternatively a USB phone set.
  Soft phones are inferior to hard phones but
  cheaper to obtain, many are available as a free
  download.

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Skype

• A proprietary protocol VOIP system built using
  Peer-to-peer (P2P) techniques.
• Free for non commercial use when using softphones
  (PC to PC).
• Offers toll access to PSTN via SkypeOut and
  SkypeIn
• From the company that created KaZaA

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free Pc-to-Pc VoIP calls

• Yahoo Messenger
• Skype
• ICQ
• MSN
• Wavigo
• Babble
• I Connect Here
• Glo Phone
• 3 W Tel
• Buddy Talk
• Pc-Telephone
• Rhino Bell
• Terra Call
• V Buzzer
• Microsoft Net Meeting

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The simplest form of VOIP is a computer-to-compute
r voice connection
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Mekkora a VoIP piac?
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Hol szaporodik a VoIP?
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Végberendezések áramellátása
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Hálózati aktív elemek áramellátása
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Segélyhívó szolgáltatások (112, 104, 105, 107)
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További lényeges kérdések

• Telefonkönyv
• Tudakozó
• Törvényes lehallgatás
• Garantált minoség

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Key issues in VoIP

• SIP
• Voice CODEC
• Packet Loss Control

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SIP (Session Initiation Protocol)

• Creation and management of a session, where a
  session is considered an exchange of data between
  an association of participants.
• Users may
• move between endpoints
• addressable by multiple names
• communicate in several different media -
  sometimes simultaneously.

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SIP

• Numerous protocols have been authored that carry
  various forms of real-time multimedia session
  data such as voice, video, or text messages. The
  Session Initiation Protocol (SIP) works in
  concert with these protocols by enabling Internet
  endpoints (called user agents) to discover one
  another and to agree on a characterization of a
  session they would like to share. For locating
  prospective session participants, and for other
  functions, SIP enables the creation of an
  infrastructure of network hosts (called proxy
  servers) to which user agents can send
  registrations, invitations to sessions, and other
  requests. SIP is an agile, general-purpose tool
  for creating, modifying, and terminating sessions
  that works independently of underlying transport
  protocols and without dependency on the type of
  session that is being established.

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SIP Functionality

• SIP is an application-layer control protocol that
  can establish, modify, and terminate multimedia
  sessions (conferences) such as Internet telephony
  calls. SIP can also invite participants to
  already existing sessions, such as multicast
  conferences. Media can be added to (and removed
  from) an existing session. SIP transparently
  supports name mapping and redirection services,
  which supports personal mobility (users can
  maintain a single externally visible identifier
  regardless of their network location).

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SIP supports five facets of establishing and
terminating multimedia communications

• User location determination of the end system to
  be used for communication
• User availability determination of the
  willingness of the called party to engage in
  communications
• User capabilities determination of the media and
  media parameters to be used
• Session setup "ringing", establishment of
  session parameters at both called and calling
  party
• Session management including transfer and
  termination of sessions, modifying session
  parameters, and invoking services.

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VoIP CODECS

• Codecs are used to convert an analog voice signal
  to digitally encoded version. Codecs vary in the
  sound quality, the bandwidth required, the
  computational requirements, etc.
• Each service, program, phone, gateway, etc
  typically supports several different codecs, and
  when talking to each other, negotiate which codec
  they will use.

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VoIP CODECS

• As an example, a Cisco ATA-186 supports these
  codecs
• G.723.1, G.711a, G.711u, G.729a
• As an example, a Cisco 7960 supports (Firmware
  P0S3-06-0-00)
• G.711a, G.711u, G.729a

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VoIP CODEC Family

• GIPS Family - 13.3 Kbps and up
• GSM - 13 Kbps (full rate), 20ms frame size
• iLBC - 15Kbps,20ms frame size 13.3 Kbps, 30ms
  frame size
• ITU G.711 - 64 Kbps, sample-based Also known as
  alaw/ulaw
• ITU G.722 - 48/56/64 Kbps ADPCM 7Khz audio
  bandwidth
• ITU G.722.1 - 24/32 Kbps 7Khz audio bandwidth
  (based on Polycom's SIREN codec)
• ITU G.722.1C - 32 Kbps, a Polycom extension,
  14Khz audio bandwidth
• ITU G.722.2 - 6.6Kbps to 23.85Kbps. Also known as
  AMR-WB. CELP 7Khz audio bandwidth
• ITU G.723.1 - 5.3/6.3 Kbps, 30ms frame size
• ITU G.726 - 16/24/32/40 Kbps
• ITU G.728 - 16 Kbps
• ITU G.729 - 8 Kbps, 10ms frame size
• Speex - 2.15 to 44.2 Kbps
• LPC10 - 2.5 Kbps
• DoD CELP - 4.8 Kbps

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• To use G.729 or G.723.1 you may need to pay a
  royalty fee!!!!!!!!!!
• this code is available for you to download for
  education purposes only!!!!!!!!!!!!

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• In VoIP networks, codecs are used to compress
  regular audio (16 bit signed linear audio,
  usually sampled at 8000Hz). Codecs are usually
  lossy'. This means that the output data does not
  have to be perfectly identical to the source data
  , it just has to sound the same when converted to
  sound.
• If your VoIP network is on an office LAN and the
  signal doesn't ever traverse a WAN connection
  (internet, VPN, DSL, etc), then compression isn't
  critical. If your VoIP signals may need to
  traverse a WAN, then you need to compress the
  signal as much as possible. This allows you to
  fit more simultaneous phone calls into a single
  WAN connection. Compression also creates smaller
  packets. Smaller packets means less audible delay
  and lower risk of packet loss.

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• Many devices offer only 1 or 2 low bit rate
  codecs, usually G.729 and one other or just
  G.729. If you have bought phones that only
  support G.729, then you have little choice.
• Some gateway providers will only allow you to
  talk to their gateway with G.729.
• A good G.729 implementation uses less bandwidth
  and less CPU power than other low bit rate codecs
  such as iLBC. G.729 uses 8kbps, iLBC uses 13kbps.
  
• Some people have observed their CPU performing up
  to 50 better when doing G.729 compression
  compared to iLBC.

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• Few phones implement iLBC (one such phone is
  Budgetone 101 and 102). Many others - Cisco 7940,
  Snom, Swissvoice - only offer G.729
• Most phones offer G.711 (ulaw/alaw) as well -
  that is actually 64kbps, eight times the
  bandwidth required by G.729. It is only for use
  on LANs.
• G.723.1 is used for similar reasons to those just
  listed, but gives the benefit of using even less
  bandwidth but with a more noticable degradation
  of sound quality.

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• Features of G.729, G.729A G.729AB Vocoder
• Compresses 8 kHz CODEC or linear audio data to 8
  kbps.
• Operates on 10ms frames with short algorithm
  delays.
• Short-term synthesis filter is based on a 10th
  order Linear Prediction (LP) filter.
• Long-term, or pitch synthesis, filter is
  implemented using the adaptive-code book
  approach.

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Hasznos linkek

• https//wiki.voip.niif.hu/index.php/VoIPSzak.C3.A
  1csk.C3.B6nyv.2FCookbook.2FTutorial