Open Source VoIP at Penn
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The University of Pennsylvania has deployed a campus-wide Voice over IP system using open source server components and protocols. The system includes SIP registrar, proxy, and presence servers using software like SER and OpenSIPS. Handsets are from Polycom. The deployment has faced challenges with features like bridged line appearances and interoperability issues, but open source allows customization. Future plans include SIP trunking and security improvements.
Open Source VoIP at Penn
- 1. Open SourceVoice over IP (VoIP) at Penn Shumon Huque University of Pennsylvania Winter 2011 Joint Techs Conference January 31st 2011, Clemson, SC, U.S.A. 1 1
- 2. Open SourceVoIP at Penn, Joint Techs, January 2011 The University of Pennsylvania is in the midst of a multi-year deployment of a campus-wide Voice over IP system based on open source components on the server side of the infrastructure and open protocols (SIP). This talk will review the architectural details, progress to date, future plans, and touch on some of the specific technical challenges we've faced. 2 2
- 3. Open SourceVoIP at Penn, Joint Techs, January 2011 Brief background • Analog Telephone system: • Verizon Centrex, over 20K lines • Old copper infra, outages, long prov time • Protocol research & testing in late 90’s • H.323 initially, later SIP (Session Initiation Protocol) • FormalVoIP project began 2005/2006 • 6,500VoIP lines so far (production) 3 3
- 4. Open SourceVoIP at Penn, Joint Techs, January 2011 Server Infrastructure • SIP Registrar & Proxy servers (iptel SER) • Voicemail servers (Asterisk) • SIP Presence servers (OpenSIPS) • PSTN gateways (cisco 3845 routers + voice cards) • In-trial: SIPTrunking (Verizon ITSP) 4 4
- 5. Open SourceVoIP at Penn, Joint Techs, January 2011 Clients • Handsets from Polycom (Soundpoint IP 321/550/650, Soundstation 6000) • Have previously used Cisco handsets (7940 and 7960) • Soft Clients: experimental, small number of users; not supported in production 5 5
- 6. Open SourceVoIP at Penn, Joint Techs, January 2011 6 6
- 7. Open SourceVoIP at Penn, Joint Techs, January 2011 7 7
- 8. Open SourceVoIP at Penn, Joint Techs, January 2011 Sampling of Features • Basic Single Line • Ring Groups • Call Hold & Transfer • Call Forward All • Call Forward Busy • Call Forward No-Ans • Call Hunt • Music on hold • Staged/timed services • Do Not Disturb • Per extensionVM dest • Caller ID block • Anonymous Rejection • Out-call notification • Distribution messages • Advanced Caller Menus 8 8
- 9. Open SourceVoIP at Penn, Joint Techs, January 2011 Web Feature Management 9 9
- 10. Open SourceVoIP at Penn, Joint Techs, January 2011 Challenges/Issues • Many bugs and interoperability issues • Timer issues, call loops, call transfer, forward, phone crashes • System tuning and scaling issues • IMAP storage of voicemail messages (for UC) • Keeping up with SER community development • BLA/SLA (Bridged/Shared Line Appearance) 10 10
- 11. Open SourceVoIP at Penn, Joint Techs, January 2011 S.E.R. Evolution 11 OpenSER Kamailio SER v2 SER SIP Router OpenSIPS Planned proxy upgrade Latest BLA fixes 2005 2008 11
- 12. Open SourceVoIP at Penn, Joint Techs, January 2011 BLA Issues • Bridged Line Appearance: multiple sets share a number; call can be picked up at one set; held; transferred to another set etc • Bugs and Interoperability issues with presence server (OpenSIPS) and handset (Polycom) • Unclear (and unfinished) technical specifications for BLA (expired Internet-drafts etc; new BLA “requirements” draft) • Deployed; backed out; debugging & repairing work going on for past 2 years • Early Jan: working reliably in our lab 12 12
- 13. Open SourceVoIP at Penn, Joint Techs, January 2011 BLA Issues • Dialogs stuck in various states (early, confirmed) -- stuck or incorrect lights on UI • Stability issues with OpenSER • Subtle interaction issues with other features (eg. call transfer, call forward, etc) • Many rounds of fixes by various involved parties (us, opensips, polycom, etc) 13 13
- 14. Open SourceVoIP at Penn, Joint Techs, January 2011 Future Enhancements • ITSP (SIP Trunking) • Security Enhancements • Secure Signalling (SIP over TLS, etc) • Secure Media (SRTP, ZRTP, etc) • Production support of Soft Clients • Automatic location tracking (public safety) • Proxy server update:“SIP Router” 3.x 14 14
- 15. Open SourceVoIP at Penn, Joint Techs, January 2011 Assessment • OpenSourceVoIP works and at large scale • But, implementing certain advanced business class telephony features is challenging • Need to be closely involved in open source development community and participate • State of maturity of protocol specs is lacking • Need strong relationships with other vendors 15 15
- 16. Open SourceVoIP at Penn, Joint Techs, January 2011 Assessment • Cost savings: no purchase or license fees • Vendor neutrality • Locally customizable, locally fixable • Ability to troubleshoot and debug better • Shared community of knowledge • Developers interested in open-standards and compatibility 16 16
- 17. Open SourceVoIP at Penn, Joint Techs, January 2011 17 Questions? Shumon Huque shuque -@- upenn.edu 17
- 18. Open SourceVoIP at Penn, Joint Techs, January 2011 Didn’t address • Organizational/Staffing issues • Project management structure • Local IT and user support issues • etc 18 18