SIP URI @ fscons 2009

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The Telecom companies is still blocking free calls. I would like to present and promote that there is techniques that will make VoIP calls as free as Emails are.
The standards that is presented is SIP URI and DNS SRV and with these combined can a company have the same VoIP & Email address.

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SIP URI @ fscons 2009

  1. 2. nobody@linux:~$ whoami Mats Karlsson [email_address]
  2. 3. Abstract The Telecom companies is still blocking free calls. I would like to present and promote that there is techniques that will make VoIP calls as free as Emails are. The standards that is presented is SIP URI and DNS SRV and with these combined can a company have the same VoIP & Email address.
  3. 4. Vision <ul><li>Email
  4. 5. Chat (XMPP, MSN)
  5. 6. Voice/Video (SIP) </li></ul><ul>One Identity </ul>All comes together to [email_address] for all types of communication!
  6. 7. The way to get there! <ul><li>DNS
  7. 8. SIP
  8. 9. XMPP </li></ul>
  9. 10. DNS <ul><li>RFC-2782, February 2000
  10. 11. BIND example... </li></ul>
  11. 12. <ul>Record format </ul><ul>An SRV record has the form: _Service._Proto.Name TTL Class SRV Priority Weight Port Target </ul>
  12. 13. <ul>Record format </ul>Service: The symbolic name of the desired service. Proto: The protocol of the desired service; usually either TCP or UDP. Name: The domain name for which this record is valid. TTL: Standard DNS time to live field. Class: Standard DNS class field (this is always IN). Priority: The priority of the target host, lower value means more preferred. Weight: A relative weight for records with the same priority. Port: The TCP or UDP port on which the service is to be found. Target: The canonical hostname of the machine providing the service. <ul>_Service._Proto.Name TTL Class SRV Priority Weight Port Target </ul>
  13. 14. Example An example SRV record might look like this: _sip._tcp.example.com. 86400 IN SRV 0 5 5060 sipserver.example.com. Another example: _sip._tcp.example.com. 86400 IN SRV 10 60 5060 sip1.example.com. _sip._tcp.example.com. 86400 IN SRV 10 20 5060 sip2.example.com. _sip._tcp.example.com. 86400 IN SRV 10 10 5060 sip3.example.com. _sip._tcp.example.com. 86400 IN SRV 20 0 5060 sip4.example.com.
  14. 15. Verify a SRV In Linux, UNIX or OS X operating systems you can use the host command to verify that the SRV record is created. host -t SRV _sip._udp.edvina.net or host -t SRV _sip._tcp.edvina.net In Windows operating systems you can verify the SRV record as follows (from the command prompt): nslookup set type=srv _sip._udp.edvina.net
  15. 16. DNS SRV usage DNS SRV records can also be used for: <ul><li>LDAP
  16. 17. Kerberos
  17. 18. XMPP
  18. 19. SMTP
  19. 20. +++ </li></ul>
  20. 21. DNS SRV impl. ex. <ul><li>Microsoft uses DNS SRV for locating services in their Active Directory components like:
  21. 22. Kerberos
  22. 23. LDAP
  23. 24. +++ </li></ul>
  24. 25. SIP <ul><li>SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996.
  25. 26. RFC-3261,June 2002
  26. 27. Asterisk example... </li></ul>
  27. 28. SIP The Session Initiation Protocol (SIP) is a signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams.
  28. 29. SIP ”Asterisk” <ul><li>In asterisk is 3 different steps needed to get this working: </li></ul><ul><li>Allow anonymous calls.
  29. 30. Inbound routing.
  30. 31. Outbound routing. </li></ul>
  31. 32. SIP ”Asterisk” Configuring Asterisk to accept anonymous and inbound URI calls: Under [general] in sip.conf make sure the following is allowed: [general] context=sip-uri ; Context for incoming calls. allowguest=yes ; Allow or reject guest calls (default ; is yes, this can also be set to 'osp'. bindport=5060 ; UDP Port to bind to (SIP standard ; port is 5060). srvlookup=yes ; Enable DNS SRV lookups on outbound ; calls.
  32. 33. SIP ”Asterisk” The context “sip-uri” is where these calls go in the extensions.conf, this is the same place that I send inbound calls from a telco or another sip peer. Under your version of the “sip-uri” context just define your dialing plan: [sip-uri] exten => s,1,NoOp(Calling remote SIP peer 123) exten => s,n,Dial(SIP/123,120,tr) exten => s,n,Congestion()
  33. 34. Test SIP URI [email_address]
  34. 35. Obstracles <ul><li>Market players
  35. 36. Technology </li><ul><li>NAT
  36. 37. SPIT
  37. 38. Security </li></ul><li>End users (adoption) </li></ul>
  38. 39. Early adopters <ul><li>Universities
  39. 40. Non profit organisations
  40. 41. “Nerds” like me... </li></ul>
  41. 42. Reference's <ul><li>en.wikipedia.org/wiki/SRV_record
  42. 43. www.dns-sd.org/ServiceTypes.html
  43. 44. www.asterisk.org
  44. 45. www.blyon.com/blog/index.php/2009/06/22/p2p-sip-uri-dialing/
  45. 46. www.enum.org </li></ul>
  46. 47. ENUM <ul><li>Map's telephone numbers to SIP URI addresses.
  47. 48. Proposed as an transition technology between POTS and ”real” IP telephony.
  48. 49. The Swedish government hasn't implemented due to lack of interest from Telco operators (Budget proposal 2006). </li></ul>
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