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Asterisk PBX: Gateway to the Future

Learn the basics of VoIP technology, its benefits, challenges, CODECS, SIP, and how the Asterisk PBX system revolutionizes communication. Get insights on voice quality, protocols, SIP addressing, and security in telecommunications.

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Asterisk PBX: Gateway to the Future

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

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

  3. 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??

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

  5. VoIP: Challenges • 3 main factors affect the quality of voice • Latency • Jitter • Packet Loss • If 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.

  6. VoIP: Quality of Voice • Quality of CODEC • give good quality low delay • Echo cancellation • 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)

  7. VoIP: CODECS • Overview of a VoIP connection: • Codecs supported by * • G.723 – 6.4kbps • G.729 – 8kbps • G.711 – 64kbps

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

  9. 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: alex@pbx.ayalanetworks.com sip:Alex Home <3001@pbx.ayalanetworks.com> sip:905-845-9430@pbx.ayalanetworks.com;user=phone sip:guest@drkangel.org

  10. VoIP: SIP Call Setup SIP User Agent Client SIP User Agent Server INVITE sip:3004@pbx.ayalanetworks.com 200 OK ACK RTP Stream BYE 200 OK 142.55.55.202 pbx.ayalanetworks.com

  11. VoIP: SIP Requests Example: INVITE

  12. VoIP: SIP REGISTER Session Initiation Protocol Request line: REGISTER sip:pbx.ayalanetworks.com SIP/2.0 Method: REGISTER Message Header Via: SIP/2.0/UDP142.55.31.239:5060;rport;branch= <omit> From: Alex <sip:3004@pbx.ayalanetworks.com> To: Alex <sip:3004@pbx.ayalanetworks.com> Contact: "Alex Ipaq" <sip:3004@142.55.31.239:5060> Call-ID: <random seed>@pbx.ayalanetworks.com CSeq: 43034 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite build 1082 Content-Length: 0

  13. VoIP: SIP INVITE Session Initiation Protocol Request line: INVITE sip:3004@pbx.ayalanetworks.com SIP/2.0 Message Header Via: SIP/2.0/UDP 142.55.55.202:5060;rport;branch=<omit> From: Alex Home <sip:3001@pbx.ayalanetworks.com>;tag=<omit> To: <sip:3004@pbx.ayalanetworks.com> Contact: <sip:3001@142.55.55.202:5060> 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:3004@pbx.ayalanetworks.com" Content-Length: 297

  14. 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

  15. VoIP: SIP Responses

  16. 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:3001@pbx.ayalanetworks.com> To: <sip:3004@pbx.ayalanetworks.com> 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

  17. 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

  18. 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

  19. 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

  20. 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

  21. Asterisk: Overview

  22. 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

  23. 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)

  24. 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

  25. Asterisk: Enum Example

  26. 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

  27. 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)

  28. Top Ten Reasons to Run Asterisk

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

  30. Number 9 Dial-an-MP3

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

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

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

  34. Number 5 You can have a 31337 answering machine.

  35. 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”.

  36. Number 3 Answer unwanted calls (ex-girlfriend) with a looping IVR “press 1 to speak to Alex…<beep>..Invalid option, please try again…”

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

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

  39. 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

  40. THANK YOU Telecom Class ‘03

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