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IP Telephony

IP Telephony. Sunday A. Folayan. Disclaimer. I am NOT an expert in VoIP technology I am NOT PRETENDING to be one. I am a user who just got interested in the technology. … and its Cool What I say may not be what it is, but how I understand it.

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IP Telephony

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  1. IP Telephony Sunday A. Folayan

  2. Disclaimer I am NOT an expert in VoIP technology I am NOT PRETENDING to be one. I am a user who just got interested in the technology. … and its Cool What I say may not be what it is, but how I understand it. Do not believe what I say wholesome, but seek your own understanding

  3. Paradigm Shift • In the beginning … geeks struggled to convert data (IP) into sound, and move it over plain old telephone system (POTS) infrastructure [using MODEMs] • Now … geeks are struggling to convert sound into data, and move it over well established IP links. [using CODECs] • Yes of Course … suits smile to the bank …

  4. So …What is IP Telephony? The packetisation and transport of classic public switched telephone system audio over an IP network. The analog audio stream is encoding in a digital format, with possible compression and filtering, before encapsulating it in IP for transport over your LAN/WAN or the public internet Infrastructure We can (safely) inter-use IP Telephony and Voice over IP (VoIP)

  5. TCP vs VoIP VoIP is not a protocol. VoIP is a collection of protocols and devices that allow for the encoding, transport and routing of audio calls over IP networks. VoIP  VoIP [P2P, Skype, Messanger] VoIP  PSTN [Net2Phone] PSTN  VoIP  PSTN [ITXC]

  6. That Simple? • No!! Imagine paying for Internet Access, by Traceroute. VoIP is incompatible with PSTNs’ charge model • PSTNs traditionally (Graham Bell Era) stuff a single call on a single cable pair … and charge for 1 pair! • PSTNs now stuff multiple calls on a single cable pair using a technique called Time Division Multiplexing (TDM) and charge as multiple pairs!! • BRI, PRI, ISDN, E1 T1 etc are all TDM technologies with diverse switching and Trunking technologies

  7. ISP2 ISP1 PSTN1 PSTN2 FXS/FXO Cisco ATA Channel Bank PRI TDM IP Phone IP RAD TDMoIP Domains POTS FXS/FXO Voice ??

  8. VoIP Building block VoIP is not built on TCP, but RTP • RTP (Real-Time Transport Protocol) • RTCP (Real-Time Control Protocol) • RTP is a UDP stream with no intelligence for QOS or resource reservation • Contains a packet number for detection of packet loss and re-sequencing of out of order packets. • Unidirectional : two streams in any call

  9. VoIP Building block • Calls are CODed to IP or DECoded from IP. • CODECS vary in sample size, usually Kbits per second • Decoding can include echo cancellation • Decoding can compensate for jitter • IP routers do not need to decode voice passing through them

  10. VoIP Building block Sample CODEC Sizes • G711alaw 64k • G711ulaw 64k • ILBC 15k • Speex 2.15 – 44.2k • Gsm 13k • G729 8k • G723 5.3 - 6.3k • Iax2 (trunked) 4k

  11. Fixed Line PSTN Mobile operator .sn Mobile Telco e164.arpa dns ENUM tree Psg.com DB based subscribers A wide choice of Providers and paths Roaming ENUM lookup +27 217 451230 Query NAPTR 200067@fwd.pulver.com PRI: +43 1 79564 Randy@psg.com Invite:100@84.201.255.254 AS5300 Freeworld Dialup Psg.com asterisk Server HP Ze5500 19343@fwd.pulver.com Sghuter@nsrc.org 19918@fwd.pulver.com TESPOK SIP Proxy Call forwarding to AS 5300

  12. Control Protocols • H323 – Complex, multiple flow, ancient • Has a large install base • Session Initiation Protocol (SIP) • New, simple, only sets up RTP streams • Cisco Skinny (Propritary) • Allows complete phone customization • MGCP (mediatrix Gateway Control Protocol) • Went the way of AMIGA. Good but Not widely deployed because it was initially close source  • IAX (Inter-Asterisk eXchange) • Simple, transverses NAT, Compessed

  13. SIP • SIP is responsible for the increased use of VoIP • Supports Video • There's lots of hardware SIP units available • Grandstream BT-101/2 • Cisco 79xx ) • Not suited for Trunking (pbx to pbx) • SIP messages are HTTP-like and readable

  14. IAX(2) • Inter Asterisk Exchange • Not many Hardware phones support IAX. • Soft Clients available for *unix/Windows • Works behind NAT • Has Trunking support built in • Very low bandwidth requirement • Built for asterisk

  15. Phones • Hard phones • Cisco 79XX’s • Grandstream BT 10X’s • Snom 100/200’s • LOTS of h.323 phones from .tw ;-) • Many other phones

  16. Phones • Soft phones • X-lite - www.xten.com (Windows) • Lipz - www.lipz4.com (Linux) • DIAX • Gnophone • Linphone - www.linphone.org (FreeBSD) • Sjphone • Lots of others

  17. Most IP phones can work Peer to Peer It is the Ability to use a PC as switch or PBX that really makes VoIP rock!! Simply loading a software PBX on a PC offers new possibilities …

  18. PBX Software Cisco Call Manager • Closed Source • 13  16 CD’s • Web Interface • Requires CCNA to setup • Needs extremely powerful Server • Leaves PRI/FXO/FXS to other devices Asterisk • Open Source • Uses industry-wide devices and equipment

  19. Asterisk is a …. • Telephony gateway (TDM - PRI,POTS) • VoIP Gateway (IP channels) • IVR system (Interactive Voice Response) • Voicemail system • Meetme Conference system • Scriptable telephony-to-anything (Perl, C, etc.) • Automatic Call distribution (ACD) system

  20. Asterisk is not … • A billing system • A CRM system • A web server or XML server (re: Cisco 79xx) • A configuration tool for VoIP devices • A voice recognition system • A USENET or email client

  21. System Requirements • No clear rule of thumb on processor size; at least 400mhz PIII recommended • Almost any version of Linux supported; RH 7.x or 8 is dev platform (9 has some issues) • Source + binaries (including sounds) are ~35m • Using complex codecs (i.e.: G.729, speex, etc.) will increase processor load dramatically • Best to have a >1.5ghz machine for multi-channel use • Linux preferred, though FreeBSD slowly starting to become stable for non-hardware channels

  22. Channel types - non-IP • POTS cards (Digium, Zapata, Voicetronix, etc.) • TDM Digital (AdTran VoFR, Digium E1/T1, etc.) • CAPI (ISDN card support for Linux ISDN driver) • USB dongle for FXS • Modem drivers for certain modems • Speaker/headphones via soundcard

  23. Configuration options • SOHO Users • FXO • Plugs straight into the phone line • Wildcard X100P Zapata cards • Internet jack • A few others • ISDN • Three Major ways of doing this with asterisk • ISDN4L • CAPI • BRI

  24. Configuration options • Business Users • BRI IDSN • Two Major ways of doing this with asterisk • CAPI • BRI • PRI ISDN • Single T1 or E1 Cards • 4 port T1 or E1 cards

  25. Basic Configuration • Load Appropriate drivers [None is needed for IP or soft phone] • Configure modules.conf • Configure either sip.conf or iax.conf • Configure extensions.conf • Start Asterisk • Make calls!

  26. Call Flow • Calls come in on channels and are then handed to the “extensions.conf” file, which is the dialplan • Dialplan contains logical sections of matches called ‘Contexts,’ and each channel sends a call into the dialplan with a context name and a dialed number • The dialplan then matches (with modified regexp’s) the number being dialed, and runs applications accordingly • Each match on the dialed number has an order of steps called ‘Priorities’, and are indicated with an integral incrementing number (BASIC-like)

  27. Regular expressions (briefly) • All regular expressions start with “_” character in dial examinations. • “X” means any number, “N” is any number other than 0 or 1 • “.” means any number of characters • Brackets represent groups, with standard “-” and “,” meanings ([1-9] or [0,1,2]) • Example: _1410985012X is the same as _1410985012[0-9]

  28. Call Flow (cont’d) [afnog] exten => 141,1,Answer exten => 141,2,Wait(2) exten => 141,3,Playback(monkeys) exten => 141,4,Goto(conference,123,1) [noc] exten => _12X,1,Playback(sorry-no-more) exten => _12X,2,Hangup

  29. Variables • ${VARNAME} is how variables are used • Variables must be declared before Booleans can be performed (gah - no null value comparitor) • Variables can be nested during setting • Exten => 123,1,SetVar(BAR=blah) • Exten => 123,2,SetVar(FOO=3) • Exten => 123,3,SetVar(NEWVAR.${FOO} = ${BAR}) • This results in ${NEWVAR.3} being set to “blah”

  30. Special Variables • ${EXTEN} - always the most important variable. This is the number that is being currently evaluated. • ${CALLERIDNUM} - the ANI (if available) of the call leg that is creating the call • Some others, less used: ${EPOCH}, ${ENV(var)}, ${CONTEXT}, ${PRIORITY}, several other descriptors of the call leg we’re processing

  31. Some Applications • Dial - connects an inbound call with some other channel. Need to specify the technology (SIP, Zap, H323, etc.) the number to be dialed, the Ring-No-Answer delay, and options (if desired) exten => 1234,1,Dial(SIP/1234,25) exten => 1234,2,Voicemail2(u1234)

  32. Some Applications (cont’d) • Playback(filename) • Plays a sound file in .gsm format • Background(filename) • Plays a sound file and gets (touch tone) input • MeetMe(conf#) • Adds the caller to a conference room (optionally muted or unmuted) [test] exten => 123,1,Background(press-a-number) exten => 123,2,Goto(1) exten => _X,1,SayDigits(${EXTEN})

  33. Some Applications (cont’d) • DISA • Lets callers from one channel get dialtone on another channel • SetMusicOnHold • You can specify .mp3 files as music on hold selections (random or sequential) • MP3Player • You can specify files or streams of .mp3 to be played to callers.

  34. Practical Uses (office) • Ditch your LD company (see prior slide) • Interconnect office PBXs at zero network cost • Get “Unified Messaging” • Give ubiquitous access to the PBX for home/traveling employees • Disaster recovery scenarios • Move phones into your IT department and away from your expensive PBX consulting firm • Eliminate adds/moves/changes as physical chores

  35. Advanced Topics • Call queues - you can build a call center with Asterisk, with various call weightings and agent logins/hot seating • Multi-ring, cascading ring with different technologies (inbound calls forward to your desk line and your cell phone - first answer gets it) • Multi-language support with same dialplan • Festival integration for voice synthesis

  36. References …. • http://www.asterisk.org/ • http://www.digium.com/ • http://www.loligo.com/asterisk/ • http://www.wwworks-inc.com/asterisk/ • http://www.xten.com/ • http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html • http://www.nznog.org/crigby-voip-intro.ppt • http://www.loligo.com/asterisk/misc/presentations/asterisk-overview.v1.0.ppt • http://docbox.etsi.org/tispan/open/enum-workshop-20040224-sophia/08.%20r%20stastny%20austria_v4.ppt • http://www.ietf.org/proceedings/03jul/slides/enum-3/enum-3.ppt • http://www.ispa.at/downloads/c8431676f72b_2003-05_ispa_enum_voip_stastny.ppt • www.voip-info.org

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