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. Do not believe what I say wholesome, but seek your own understanding

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 …

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)

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]

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

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

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

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

VoIP Building block Sample CODEC Sizes –G711alaw64k –G711ulaw64k –ILBC15k –Speex2.15 – 44.2k –Gsm13k –G7298k –G k –Iax2 (trunked)4k

Freeworld Dialup Psg.com asterisk Server Psg.com DB based subscribers Query NAPTR Fixed Line PSTN Mobile Telco AS5300 Call forwarding to AS 5300 PRI: Roaming Mobile operator.sn HP Ze5500 ENUM lookup TESPOK SIP Proxy e164.arpa dns ENUM tree A wide choice of Providers and paths

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

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

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

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

Phones Soft phones –X-lite - (Windows) –Lipz- (Linux) –DIAX –Gnophone –Linphone- (FreeBSD) –Sjphone –Lots of others

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 …

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

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

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 client

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

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

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

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

Basic Configuration 1.Load Appropriate drivers [None is needed for IP or soft phone] 1.Configure modules.conf 2.Configure either sip.conf or iax.conf 3.Configure extensions.conf 2.Start Asterisk 3.Make calls!

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)

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: _ X is the same as _ [0-9]

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

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”

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

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,Voic 2(u1234)

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})

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.

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

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

References … overview.v1.0.ppthttp:// overview.v1.0.ppt sophia/08.%20r%20stastny%20austria_v4.ppthttp://docbox.etsi.org/tispan/open/enum-workshop sophia/08.%20r%20stastny%20austria_v4.ppt _ispa_enum_voip_stastny.ppthttp:// 05_ispa_enum_voip_stastny.ppt