Asterisk PBX: VoIP’s gateway to the future By Alex Ayala For

Asterisk PBX: VoIP’s gateway to the future By Alex Ayala For Telecom class of 2003

Agenda Introduction to VoIP Benefits Challenges CODECS Session Initiation Protocol Asterisk PBX Demonstration

What is VoIP? Based on packet switching technology using Internet as transport Opposed to the traditional circuit switching technology, which dominates the Public Switched Telephone Network (PSTN) Driven by low cost; flat-rate billing So why haven’t we switch to VoIP?

VoIP: Benefits Integration of Data & Voice Simplification Less equipment management Network Save on Bandwidth (silence suppression) Cost Efficiently Reduction Bypass PSTN toll fees

VoIP: Challenges 3 If main factors affect the quality of voice Latency Jitter Packet Loss cost is the only criteria Managers/Administration would be only ones who wouldn’t mind bad voice quality. Employees won’t compromise quality to reduce company’s bills.

VoIP: Quality of Voice Quality of CODEC Echo cancellation give good quality low delay 2 wire - 4 wire PBX (hybrid circuit used for conversion) if delay 10mS echo is notice Delay Total Delay ( 200mS one-way; talkers overlap ) Jitter ( variable packet arrival ) Delay Management Prioritize (RSVP) Packet replay (Jitter buffer) Segmenting data packets (exit router faster)

VoIP: CODECS Overview of a VoIP connection: Codecs supported by * G.723 – 6.4kbps G.729 – 8kbps G.711 – 64kbps Voice ADC DAC Voice Compression Algorithm Decompress Algorithm Assembling RTP/UDP Dissembling RTP/UDP

VoIP: Protocols RSVP (Resource ReSerVation Protocol) RTP (Real Time Protocol) RTCP (Real Time Control Protocol) SIP (Session Initiation Protocol) SDP (Session Description Protocol)

VoIP: SIP Addressing Uses Internet URLs Supports both Internet and PSTN addresses General form is name@domain To complete a call, needs to be resolved down to User@Host Examples: sip: [email protected] sip:Alex Home [email protected] sip:[email protected];user phone sip:[email protected]

VoIP: SIP Call Setup SIP User Agent Client SIP User Agent Server INVITE sip:[email protected] 200 OK ACK RTP Stream BYE 200 OK 142.55.55.202 pbx.ayalanetworks.com

VoIP: SIP Requests Example: INVITE Method Description REGISTER Used by client to register a particular address with the SIP server INVITE A session is being requested to be setup using a specified media ACK Message from client to indicate that a successful response to an INVITE has been received BYE A call is being released by either party CANCEL Cancels any pending requests. Usually sent to a Proxy Server to cancel searches OPTIONS A Query to a server about its capabilities

VoIP: SIP REGISTER Session Initiation Protocol Request line: REGISTER sip:pbx.ayalanetworks.com SIP/2.0 Method: REGISTER Message Header Via: SIP/2.0/UDP 142.55.31.239:5060;rport;branch omit From: Alex sip:[email protected] To: Alex sip:[email protected] Contact: "Alex Ipaq" sip:[email protected]:5060 Call-ID: random seed @pbx.ayalanetworks.com CSeq: 43034 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite build 1082 Content-Length: 0

VoIP: SIP INVITE Session Initiation Protocol Request line: INVITE sip:[email protected] SIP/2.0 Message Header Via: SIP/2.0/UDP 142.55.55.202:5060;rport;branch omit From: Alex Home sip:[email protected] ;tag omit To: sip:[email protected] Contact: sip:[email protected]:5060 Call-ID: omit @142.55.55.202 CSeq: 23277 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite build 1088 Proxy-Authorization: Digest username "3001",realm "asterisk",nonce 4c3e876b, response “ hash ”,uri "sip:[email protected]" Content-Length: 297

VoIP: SDP Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 3001 173802875 173802875 IN IP4 142.55.55.202 Session Name (s): X-Lite Connection Information (c): IN IP4 142.55.55.202 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8000 RTP/AVP 0 8 Media Attribute (a): rtpmap:0 pcmu/8000 Media Attribute (a): rtpmap:8 pcma/8000 Media Attribute (a): rtpmap:3 gsm/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): rtpmap:97 speex/8000

VoIP: SIP Responses Description Examples 1xx Informational – Request received, continuing to process request. 180 Ringing 100 Trying 2xx Success – Action was successfully received, understood and accepted. 200 OK 3xx Redirection – Further action needs to be taken in order to complete the request. 300 Multiple Choices 302 Moved Temporarily 4xx Client Error – Request contains bad syntax or cannot be fulfilled at this server. 401 Unauthorized 408 Request Timeout 5xx Server Error – Server failed to fulfill an apparently valid request. 503 Service Unavailable 505 Version Not Suported 6xx Global Failure – Request is invalid at any server. 600 Busy Everywhere 603 Decline

VoIP: SIP Responses (cont) Required Fields: SIP/2.0 200 OK Via: SIP/2.0/UDP 142.55.55.202:5060 From: Alex Home sip:[email protected] To: sip:[email protected] Call-ID: omit @142.55.55.202 CSeq: 23278 INVITE These are copied from the request corresponding to 200 OK To and From are NOT swapped CSeq is incremented by 1

VoIP: SIP Routing VIA headers are used for routing SIP messages Requests Request Initiator puts address in VIA header Responses Response initiator copies request VIA header

VoIP: SIP Security ENCRYPTION SIP offers various approaches End 2 end encryption Hob by hop encryption AUTHENTICATION Proxies might require auth Responds to INVITE with 407 proxy auth req. Client re-INVITE with Proxy Authorization header UAS/Registrars might require auth Responds to INVITE with 401 unauthorize Client re-INVITE with Authorization header

Asterisk:What is it? A complete PBX software for Linux platform developed by Digium (M.S.) Does PBX call switching, CODEC translation, and various applications Open Source under GNU license

Asterisk: Applications Voicemail Dial an interface (ZAP, SIP, IAX, etc) Conference Bridging ACD Queues (great for Call centres) IVR ( press “1” if you know the ext) DB operations ENUMlookup AGI (asterisk gateway interface, like CGI) For advance scripting

Asterisk: Overview VoIP EnumLookup VoIP Voicemail PSTN ASTERISK PBX Queue Analog Phones Conference Bridging PSTN Analog Phones

Asterisk: Call Logic Asterisk uses a State Machine to determine what to do with a Call Context : The Origin of the call (SIP, PSTN, etc) Extension: The number Dialed by user Priority: A counter that orders a sequence of commands

Asterisk: Call Logic Example A user dials 3001, which is extension for Voicemail Central. The user is define in context local extensions.conf [local] exten 3001,1,Voicemailmain2 A sip user (4001) dials 1001 which is an analog phone (Zap/1), and drop in voicemail if unavailable (no one answers for 30 secs) sip.conf [4001] Username 4001 Context from-sip extensions.conf [from-sip] exten 1001,1,Dial(Zap/1,30) exten 1001,2,Voicemail2(u1001)

Asterisk: ENUM A PSTN user wants to call a SIP user? Only have a dialpad. How to dial a URI? ENUM. Creates a global directory which map telephone number to sip address (or email ). DNS lookup (E.164 - URIs) E.164 queries are formed as reversed dot-separated digits and attach the enum.domain.tld at the end (usualy e164.arpa) 905-845-9430 0.3.4.9.5.4.8.5.0.9.e164.arpa

Asterisk: Enum Example ENUM IN NAPTR 0.3.4.9.5.4.8.5.0.9.e164.arpa. ? PSTN: 905-845-9430 GW w/ Enum resolution sip:[email protected] INVITE: sip:[email protected] sip:[email protected]

Asterisk: IAX Inter-Asterisk eXchange used by Asterisk as an alternative to SIP, H.323, etc Supports PKI-style security and trunking When trunking, it allocates BW in used only Quality is similar to SIP, but as connections increase IAX (in trunk mode) becomes better. Versions: IAX and IAX2

Asterisk: IAX (cont) IAX is NAT/PAT transparent IAX2 trunking triples per megabyte 100 calls/MB (with G.729) Over 1000 iaxtel registered users (like FWD)

Top Ten Reasons to Run Asterisk

Number 10 Convenient, unambiguous single non-alphanumeric abbreviation: *

Number 9 Dial-an-MP3

Number 8 Can call you 5 minutes into a blind date as 'emergency exit'

Number 7 Only way to build a call center on your laptop

Number 6 Teleconferencing with your friends allows you to be more lazy/unsocial than you already are

Number 5 You can have a 31337 answering machine.

Number 4 Finally you can tell telemarketers , “all representatives of our household are busy attending other telemarketers, your call will be answer in order of received”.

Number 3 Answer unwanted calls (exgirlfriend) with a looping IVR “press 1 to speak to Alex beep .Invalid option, please try again ”

Number 2 Have screaming parents,siblings,etc after they can’t call long distance, Password protected.

Number 1 Why settle for being just another webmaster, hostmaster, or postmaster when you too can be an astmaster like me!

Asterisk: Demo 2 Asterisk servers 4 Sip clients , 4 local phones (2 in each server) IAX2 trunk between servers Both will act as sip proxies Server A is connected to PSTN via FXO Using ENUM for least cost routing

ASTERISK PBX Host: Home Phone Line connected to FXO card PSTN SERVER A SIP CLIENTS @School SIP CLIENTS @Home 4001@school 3001@home INTERNET 3002@home 1001 4002@school 2001 1002 Analog phones connected to FXS card ASTERISK PBX Host: School 2002 SERVER B Analog phones connected to FXS card

THANK YOU Telecom Class ‘03