The document discusses VoIP (Voice over Internet Protocol), benefits and challenges of VoIP including quality of voice and codecs, protocols used like SIP and RTP, and provides an example of Asterisk call logic and demonstration. It explains that VoIP uses internet packet switching instead of traditional circuit switching, has benefits like cost savings and integration of data and voice. However, quality of voice depends on factors like codec used, delay, jitter. Protocols like SIP are used for call setup and RTP for media transmission. The document concludes with a demo of an Asterisk PBX with SIP clients, local phones and an IAX2 trunk between servers, showing least cost routing using ENUM.

1. Asterisk Open Source PBX Presented by: WebKul India

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: 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)

6. VoIP: CODECS Codecs supported by * G.723 – 6.4kbps G.729 – 8kbps G.711 – 64kbps

7. VoIP: Protocols RTP (Real Time Protocol) SIP (Session Initiation Protocol) SDP (Session Description Protocol)

8. What is Voice over IP (VoIP)? Internet/ Private IP Network ) ) ) 1010101000010100101010101010010101010100101010001001 IP Packet 1010101000010100101010101010010101010100101010001001 IP Packet 1010101000010100101010101010010101010100101010001001 1010101000010100101010101010010101010100101010001001 ( ( (

9. VoIP Deployments Phone to Phone is mainly provided by Service Provider or Private Network (E.g. Singtel’s 019) 1998 IP Phone to IP Phone is mainly provided by Service Provider or self Managed (E.g. Free World Dial) 2002 post 2003 PC(Web) to Phone is mainly provided by Service Provider (E.g. Web2Phone) 1999 Wireless IP Phone to Wireless IP Phone will be provided by whom?

10. 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)

11. SIP — the Session Initiation Protocol SIP is a signalling protocol – it does not carry the ‘meat’ of conversations SIP finds people, sets up calls, and rings phones SIP allows callers to agree on data format (codecs, etc.) Kinds of SIP servers : Proxy/Redirect (relays requests) Registrar (keeps lists of users and where they/their phones are, so they can be called) Asterisk does both (discussed later)

12. 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