T.Gy. Intrernetes médiakommunikáció. 2009.05. 04. 1 Internetes médiakommunikáció VoIP Takács...

T.Gy. Intrernetes médiakommunikáció. 2009.05. 04.

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Internetes médiakommunikáció

VoIP

Takács György 10. előadás2009. 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-

computer 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 minőség


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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/VoIP#Szak.C3.A1csk.C3.B6nyv.2FCookbook.2FTutorial

https://wiki.voip.niif.hu/index.php/VoIP#Szak.C3.A1csk.C3.B6nyv.2FCookbook.2FTutorial



T.Gy. Intrernetes médiakommunikáció. 2009.05. 04. 1 Internetes médiakommunikáció VoIP Takács...

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