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TEL500-Voice Communications Session initiation protocol improvement using inter-asterisk exchange

TEL500-Voice Communications Session initiation protocol improvement using inter-asterisk exchange Devesh Mendiratta & Sameer Deshmukh MS-Telecommunication State University of New York Institute of Technology. Introduction. VOIP Network VOIP Protocols SIP ( Session initiation protocol)

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TEL500-Voice Communications Session initiation protocol improvement using inter-asterisk exchange

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  1. TEL500-Voice Communications Session initiation protocol improvement using inter-asterisk exchange Devesh Mendiratta & Sameer Deshmukh MS-Telecommunication State University of New York Institute of Technology

  2. Introduction • VOIP Network • VOIP Protocols • SIP (Session initiation protocol) • IAX (inter-asterisk exchange Protocol)

  3. Introduction cont. • Codecs • Bandwidth Utilization over VoIP Networks • Theoretical bandwidth calculation • Comparison Practical bandwidth calculation

  4. VOIP Protocols • Centralized • Distributed • H.323 • SIP • IAX

  5. VOIP Architecture Models

  6. SIP Protocol • Session Layer Protocol • “Request and Response” Mechanism • Sessions & Video data between two endpoints • File types & formats • After establishing a session – responsibility of average flow transfer is delegated to RTP i.e. Transport layer • Dynamically allocation of port (10000 to 20000)

  7. IAX Protocol • Session layer protocol • Provides control & VoIP networks • Point-to-point • Media & signaling protocol • Multiplexing of signaling • UDP port 4569 • Trunked IAX

  8. Codecs • Analog voice is converted to a digital signal • Then it is carried across the Internet. • Examples based on compression level • G.711 • G.726 • G.729A • GSM • iLBC • Speex • MP3

  9. Calculation : Theoretical Value • Codec bit rate = (codec sample size)/(codec sample interval) • Packets per seconds = (codec bit rate)/(voice payload size) • Total packet size = L2 + IP + UDP + L5 + voice payload size • The BW required for n conversations full duplex: BWn = BW x n x 2

  10. Example : SIP using codec G.711 • CSI = 10ms • CSS = 80 bytes • VPS = 160 bytes • BW30calls = 87.2 Kbps x 30 x 2 = 5232 Kbps

  11. Calculation : Theoretical Value

  12. Practical Value • Vyatta installation • Configuring interfaces & static routs on Vyatta • Installing CentOS 5.2 OS on servers Asterisk 1 & 2 • Installing Asterisk PBX on servers 1 & 2 • Installing Wireshark & Unsniff sniffer on the machine • SIP & IAX extensions settings • Dial plan configuration

  13. BW SIP, G.711, 30 CALLS

  14. BW SIP, GSM, 30 CALLS

  15. Comparison Table

  16. Conclusion • SIP & G.711 codec => very good quality of voice • highest consumption of BW • IAX & GSM codec => lowest consumption of BW • high traffic – distortion • IAX & G.711 codec => ideal for power • traffic level isrelatively high • requires high bandwidth • SIP & GSM codec => ideal for plans that do not support IAX

  17. Resources • ^http://ofps.oreilly.com/titles/9781449332426/asterisk-UnderstandingVoIP.html • ^http://en.wikipedia.org/wiki/Category:VoIP_protocols • ^http://www.cisco.com/application/pdf/en/us/guest/tech/tk587/c1506/ccmigration_09186 a008012dd36.pdf

  18. ? Thank You Any Questions Undergrad ???

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