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  1. 1. CS760S Wireless Mesh for Voice Communication Presented by: Vishesh Kumar (2001431) Rohan Rai (2001426)
  2. 2. Servers <ul><li>Asterisk </li></ul><ul><li>It is a complete PBX in software. </li></ul><ul><li>Voicemail services with Directory. </li></ul><ul><li>Call Conferencing. </li></ul><ul><li>Interactive Voice Response. </li></ul><ul><li>Call Queuing. </li></ul><ul><li>Three-way calling. </li></ul><ul><li>Caller ID services. </li></ul><ul><li>SIP, H.323 (as both client and gateway), </li></ul><ul><li>MGCP (call manager only). </li></ul>
  3. 3. Clients <ul><li>SJPhone (configured and used). </li></ul><ul><li>Dante Diax’s Phone. </li></ul><ul><li>X-Pro. </li></ul>
  4. 4. Protocols <ul><li>IAX2 (Used in inter asterisk communication) </li></ul><ul><li>SIP (Used on clients) </li></ul><ul><li>H.323 </li></ul>
  5. 5. IAX2 <ul><li>The Inter-Asterisk eXchange ( IAX ) protocol provides control and transmission of streaming media over Internet Protocol (IP) networks. The IAX revision 2 protocol is used by the Asterisk VOIP PBX as an alternative to SIP, H.323, etc. when connecting to other devices that support IAX (Here it is used for communication between two asterisk servers). </li></ul>
  6. 6. SIP <ul><li>The Session Initiation Protocol (SIP) is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. (Here it is used to communicate between client and server) </li></ul>
  7. 7. SJPhone Client 1 SJPhone Client 2 SJPhone Client 3 SJPhone Client 4 Asterisk Server 2 Asterisk Server 1 IAX2 Protocol Sip Protocol Sip Protocol
  8. 8. Incorporation of Roaming Facility <ul><li>Roaming  irrespective of the server a client should be able to use the voip facilities (Number of the client remains same) </li></ul><ul><li>A client can register to any of the server </li></ul><ul><li>The number of the client should remain unchanged </li></ul>
  9. 9. DUNDi (Distributed Universal Number Discovery) <ul><li>It is a peer-to-peer system for locating Internet gateways to telephony services. </li></ul><ul><li>It is fully-distributed </li></ul><ul><li>It has the ability to arbitrarily add new extensions, gateways and other resources to a trusted web of communication servers, where any adds, moves, changes, failures or new routes are automatically absorbed within the cloud with no additional configuration </li></ul>
  10. 10. Server 13 7 th Floor Server 12 6 th Floor Server 11 4 th Floor Server 10 3 rd Floor SERVER 9 2 nd Floor SERVER 8 1 st Floor SERVER 7 Gnd Floor SERVER 3 4 th Block SERVER 2 5 th Block SERVER 6 1 st Block SERVER 5 2 nd Block SERVER 1 6 th Block SERVER 4 3rd Block MAIN BUILDING BLUEPRINT OF THE DISTRIBUTION OF ASTERISK SERVER
  11. 11. Bandwidth Measurement                     27.7 Kbps 15 Kbps iLBC 31.5 Kbps 16 Kbps G.728 47.2 Kbps 24 Kbps G.726 55.2 Kbps 32 Kbps G.726 20.8 Kbps 5.3 Kbps G.723.1 21.9 Kbps 6.4 Kbps G.723.1 31.2 Kbps 8 Kbps G.729 87.2 Kbps 64 Kbps G.711 NEB BR Codec
  12. 12. erlang is a statistical measure of telecommunication traffic used in telephony It is the number of call hours during peak time. E.g. (For Main Campus) Erlang = 1200*60 (Peak calls / hr)+1/60(Avg time in hrs) = 1200 150 150 ~ 100 Erlang per server 3 2 13 Proposed no. of servers 450 300 1200 Erlang 90 90 60 Avg. time duration 300 200 1200 Peak calls per minute 300 200 1500 No. of Phones New Campus Old Campus Main Campus
  13. 13. References <ul><li>http://www.asterisk.org/ </li></ul><ul><li>http:// www.sjlabs.com/sjp.html </li></ul><ul><li>http://voip-info.org/wiki-asterisk+introduction </li></ul><ul><li>http://www.voip-info.org/wiki-IAX </li></ul><ul><li>http://www.voip-info.org/wiki-SIP </li></ul>
  14. 14. Thank You