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.

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.

5. Chat (XMPP, MSN)

8. SIP

9. XMPP

11. BIND example...

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

17. Kerberos

18. XMPP

19. SMTP

20. +++

22. Kerberos

23. LDAP

24. +++

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.

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]

37. SPIT

40. Non profit organisations

41. “Nerds” like me...

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

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).