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Global Asterisk Voice over IP Implementation

Global Asterisk Voice over IP Implementation. Kirk Wilson Director of IT kirk.wilson@shipsoffice.org 8-501-1000. Goals. Connect existing telephone infrastructure Cut down on long distance cost Improve internal communications Provide easy remote access with soft phones

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Global Asterisk Voice over IP Implementation

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  1. Global Asterisk Voice over IP Implementation Kirk Wilson Director of IT kirk.wilson@shipsoffice.org 8-501-1000

  2. Goals • Connect existing telephone infrastructure • Cut down on long distance cost • Improve internal communications • Provide easy remote access with soft phones • Controlled bandwidth use of VoIP services • Flexibility in providing future telephone services • Build a globally scalable configuration

  3. Why not Skype? • Controlled/configured by end-user, not IT • Every user needs computer • Only talks to other Skype clients or through Skype out at a fee • Not terribly bandwidth conservative • Uses all available bandwidth to provide best call clarity by default • Will quickly saturate a 256k satellite link, especially with multiple users • Can not trunk multiple calls

  4. Codec Bandwidth Comparison • Skype • 24k – 128k bps (may double with "station" mode) • In super node mode it may saturate up to 100Mb/s • G.711 (uLaw) 66 kb/s per call (+16 kb/s overhead) • G.729 9.6 kb/s per call (+20 kb/s overhead) • GSM 14.7 kb/s per call (+20 kb/s overhead) • LPC10 6.1 kb/s per call (+16 kb/s overhead) • SPEEX 17.4 kb/s per call (+20 kb/s overhead)

  5. Asterisk Overview • Premier open source PBX software • Developed to run on Linux • Can interconnect most existing VoIP/Digital/Analog technologies • Seamless Protocol/Codec translation • Can be deployed around existing PBX systems • Supports IAX which requires only a single port • Scalable from small to large systems

  6. Global Design Issues • Numbering Plan • Office Interconnects System designed to scale beyond OM Ships to all of OM

  7. Numbering Plan Requirements • Easy to remember • Integrate with existing varied PBX’s • Simplify call routing/interconnects

  8. Numbering Plan 8-YYY-XXXX • 8 – Asterisk access code • “Dial 8 for an asterisk line” • YYY – Office code • Can handle up to 999 OM offices • XXXX – Local extension • Can handle up to 9,999 extensions per office

  9. Numbering Plan This plan allows for easy integration into the already established telephone systems deployed at our existing offices and ships • 8-500-XXXX Mosbach HQ Office • 8-501-XXXX Carlisle Office • 8-502-XXXX Doulos • 8-503-XXXX Logos Hope

  10. Office Interconnects Requirements • Simple, easy to maintain • Scalable • Secure • Bandwidth efficient

  11. Office interconnects ENUM w/ RSA – Global Directory • Interoffice link configuration stored in one standard DNS zone. (enum.om.org) • Only one DNS record per office. • Office code assignments and DNS can be centrally managed. (ICT?) • New office: one new NAPTR record.

  12. Office interconnects ENUM w/ RSA – Local Server • Local extensions locally managed • Local changes affect only the local server • Single, unchanging, globally reusable context for each: • Inbound VoIP calls • Outbound VoIP calls • New office: No changes necessary

  13. Office interconnects ENUM w/ RSA – Connections • No traffic until a call is made • Calls are authenticated using a single “om-enum” RSA keypair

  14. .enum.om.org DNS Server Internet ENUM Result: 0.0.0.1.0.0.5.enum.om.org. IN NAPTR 100 10 “u” “E2U+IAX2” “!^\\+(.*)$!iax2:om-enum@voip.mosbach.omships.org/\\1!” Mosbach Asterisk strips office code Dials extension: 1000 Mosbach Asterisk sees inbound call to: 500-1000 ENUM Lookup: NAPTR 0.0.0.1.0.0.5.enum.om.org. Asterisk matches: _8XXXXXXX PBX passes call to Asterisk Call Completed DNS Server Matches: NAPTR 0.0.0.1.0.0.5.enum.om.org. to wildcard record NAPTR *.0.0.5.enum.om.org. User dials: 8-500-1000 Asterisk Dial: Dial(IAX2/om-enum@voip.mosbach.omships.org/5001000) RSA Public/Private Key Authentication An example ENUM Call

  15. Satellite Considerations • Problem: Severe bandwidth constraints • 256kbit up/down limit • Easily overloaded and render useless • Solution: Route all calls via Carlisle • Translate all calls to G.729 • Limit total number of inbound/outbound calls • Trunk calls for significant bandwidth savings

  16. Additional Features • Asterisk-less offices • Trogir • Date/time announcements • Time zone enumeration • Interactive voice menus • Voicemail

  17. Future possibilities • Faxing • Conferences • Worldwide VoIP to local PSTN gateways • Doulos -> South Africa PSTN phone • Only the cost of a local call • Call following and Call groups • Non-OM user soft phone access • OM Standard Asterisk Server (?)

  18. Ongoing Issues/Problems • Faxing • Can’t fax over a lossy voice codec • No support for T.31 • spandsp helps, but isn’t perfect • Authentication • Single RSA keypair for every office • Encryption • Asterisk current can not encrypt RSA authenticated channels

  19. Questions?

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