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SIP deployment in LEARN

This project highlights the deployment of Session Initiation Protocol (SIP) and VoIP services at the Lanka Educational and Research Network (LEARN) and the University of Peradeniya. Key implementations include integration of an open-source PBX using Asterisk, Direct Inward Dialing capabilities from PSTN, and efficient call routing through SER. The solution utilizes Cisco hardware and various software components to enable secure and effective communication among users, while also providing online usage tracking for individual accounts.

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SIP deployment in LEARN

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  1. Nimal Ratnayake <nimalr@learn.ac.lk> Technical Manager, Lanka Educational and Research Network (LEARN)Senior Lecturer, Department of Electrical & Electronic Engineering, University of Peradeniya SIP deployment in LEARN

  2. SIP at U of Peradeniya • Implemented sip.edu • Integrated PBX and SIP based VoIP • Direct Inward Dialing from PSTN • SER for call routing • Asterisk for PBX features • ENUM lookup

  3. SIP at Peradeniya (2)‏

  4. Components used • Mostly Open Source SW • SER and Asterisk • Cisco 3640 w/ ISDN PRI link to PBX as gateway • Ericsson MD100 PBX with MFC R2 link to PSTN • OpenLDAP for campus directory • FreeRARIUS for authentication • MySQL for SER databases and accounting • Perl Scripts for LDAP Lookup/Call Logging • Cisco, Grandstream, LinkSys and x-lite UAs

  5. Accounting • Use MySQL for storing call logging from PBX and Cisco gateway • Two simple daemons written in Perl • Capture text call logging information from PBX • Capture Radius accounting packets from gateway • Both PBX and VoIP calls go to the same user account • Users can check their usage on line

  6. SIP in LEARN • Call routing between LEARN sites • SER for call routing • Number format • 99 access code • 2 or 3 digit institution code • 4 or 3 digit extension number

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