The document discusses the potential for making VoIP calls as free as emails by utilizing SIP URI and DNS SRV standards to unify communication identities across various platforms. Detailed examples of SRV record formats and the Asterisk configuration for enabling anonymous calls are provided, emphasizing the ease of integration. It also highlights the lack of adoption by telecom companies and government institutions in Sweden due to disinterest and market obstacles.

2. nobody@linux:~$ whoami Mats Karlsson [email_address]

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.

4. Vision Email

5. Chat (XMPP, MSN)

6. Voice/Video (SIP) One Identity All comes together to [email_address] for all types of communication!

7. The way to get there! DNS

8. SIP

9. XMPP

10. DNS RFC-2782, February 2000

11. BIND example...

12. Record format An SRV record has the form: _Service._Proto.Name TTL Class SRV Priority Weight Port Target

13. Record format 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. _Service._Proto.Name TTL Class SRV Priority Weight Port Target

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.

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

16. DNS SRV usage DNS SRV records can also be used for: LDAP

17. Kerberos

18. XMPP

19. SMTP

20. +++

21. DNS SRV impl. ex. Microsoft uses DNS SRV for locating services in their Active Directory components like:

22. Kerberos

23. LDAP

24. +++

25. SIP SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996.

26. RFC-3261,June 2002

27. Asterisk example...

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.

29. SIP ”Asterisk” In asterisk is 3 different steps needed to get this working: Allow anonymous calls.

30. Inbound routing.

31. Outbound routing.

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.

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()

34. Test SIP URI [email_address]

35. Obstracles Market players

36. Technology NAT

37. SPIT

38. Security End users (adoption)

39. Early adopters Universities

40. Non profit organisations

41. “Nerds” like me...

42. Reference's en.wikipedia.org/wiki/SRV_record

43. www.dns-sd.org/ServiceTypes.html

44. www.asterisk.org

45. www.blyon.com/blog/index.php/2009/06/22/p2p-sip-uri-dialing/

46. www.enum.org

47. ENUM Map's telephone numbers to SIP URI addresses.

48. Proposed as an transition technology between POTS and ”real” IP telephony.

49. The Swedish government hasn't implemented due to lack of interest from Telco operators (Budget proposal 2006).