Method and apparatus for enhanced internet telephony

• US 20080181375A1
• Filed: 02/14/2008
• Published: 07/31/2008
• Est. Priority Date: 10/15/2003
• Status: Abandoned Application

• Current Assignee: Vonage America Incorporated (Telefonaktiebolaget LM Ericsson)
• Original Assignee: Vonage Holdings Corporation (Telefonaktiebolaget LM Ericsson)
• Inventors: Holder, Louis, Citron, Jeffrey
• Application Number:
  US12/071,005
• Publication Number:
  US 20080181375A1
• Time in Patent Office: Days
• US Class Current: 379/93.01
• CPC Class Codes: H04L 61/255 Maintenance or indexing of mapping tables
• CPC Class Codes: H04L 61/2564 for a higher-layer protocol, e.g. for session initiation protocol [SIP]
• CPC Class Codes: H04L 65/1026 at the edge
• CPC Class Codes: H04L 65/1069 Session establishment or de-establishment
• CPC Class Codes: H04L 65/1104 Session initiation protocol [SIP]

Abstract

A method and apparatus for enhanced Internet telephony ensures that communication between a source and destination is not interrupted by common network address translation. According to one aspect of the invention, communication may continue through a router that employs network address translation.

Claims

1-7. -7. (canceled)

8. An internet telephony system configured to use Session Initiation Protocol (SIP) signaling to setup a communication of streaming packets, the internet telephony system comprising:

a relay configured to relay streaming packets of the communication between a caller and a call destination;

a server configured to receive, process and transmit SIP signaling messages to setup the communication between the caller and the call destination and to select the relay to use for the communication, the selection being based at least on the quality of the communication.

9. The internet telephony system of claim 8, wherein the server is configured to select the relay based in part on the geographic location of the caller.

10. The internet telephony system of claim 8, wherein the server is configured to select the relay based in part on the geographic location of the call destination.

11. The internet telephony system of claim 8, wherein the server is configured to select the relay to decrease the latency of the communication.

12. The internet telephony system of claim 8, wherein the server is configured to select the relay to decrease the travel time of the communication.

13. The internet telephony system of claim 8, wherein the server is configured to select the relay to limit the geographical area traveled by the streaming packets of the communication.

14. The internet telephony system of claim 8, wherein the relay is associated with a point-of-presence geographically separated from other points-of-presence.

15. The internet telephony system of claim 8, wherein the server is configured to select the relay based on a SIP Invite message.

16. The internet telephony system of claim 8, wherein said server is a pre-proxy server.

17. The internet telephony system of claim 8, wherein said streaming packet protocol is the Real Time Transport Protocol (RTP).

18. The internet telephony system of claim 8, wherein the relay is a RTP relay.

19. The internet telephony system of claim 8, wherein the server is separated from the relay.

20. A method of providing internet service, the method comprising:

providing a server configured to receive, process and transmit Session Initiation Protocol (SIP) signaling messages;

receiving a signaling message originating from a caller requesting a communication to a call destination;

selecting a relay for use during the communication based at least on the quality of the communication;

relaying streaming packets of the communication via the selected relay between the caller and the call destination.

21. The method of claim 20, wherein the selecting the relay is based in part on the geographic location of the caller.

22. The method of claim 20, wherein the selecting the relay is based in part on the geographic location of the call destination.

23. The method of claim 20, wherein the selecting the relay is based in part on improving the latency of the communication.

24. The method of claim 20, wherein the selecting the relay is based in part on improving the travel time of the communication.

25. The method of claim 20, wherein the selecting the relay is based in part on limiting the geographical area traveled by the steaming packets of the communication.

26. The method of claim 20, wherein the relay is associated with a point-of-presence geographically separated from other points-of-presence.

27. The method of claim 20, wherein the SIP signaling message originating from the caller is a SIP Invite.

28. The method of claim 20, wherein the server operates as a pre-proxy server.

29. The method of claim 20, wherein the streaming packets are Real Time Transport Protocol (RTP) packets.

30. The method of claim 20, wherein the selected relay is a RTP relay.

31. The method of claim 20, wherein the server is separated from the relay.

32. An internet telephony server for setting up a communication of streaming packets, the server configured to receive and process a Session Initiation Protocol (SIP) signaling message originated from a caller requesting a communication of streaming packets to a call destination, wherein the server is configured to process information in the SIP signaling message to select a relay from a plurality of relays available to relay streaming packets of the communication between the caller and the call destination, wherein-the server makes the selection based at least on the quality of the communication.

33. The server of claim 32, further configured to select the relay based in part on the geographic location of the caller.

34. The server of claim 32, further configured to select the relay based in part on the geographic location of the call destination.

35. The server of claim 32, further configured to select the relay to decrease the latency of the communication.

36. The server of claim 32, further configured to select the relay to decrease the travel time of the communication.

37. The server of claim 32, further configured to select the relay to limit the geographical area traveled by the steaming packets of the communication.

38. The server of claim 32, wherein each relay is associated with a point-of-presence geographically separated from other points-of-presence.

39. The server of claim 32, wherein the signaling message is a SIP Invite.

40. The server of claim 32, wherein the server operates as a pre-proxy server.

41. The server of claim 32, wherein the streaming packets are Real Time Transport Protocol (RTP) packets.

42. The server of claim 32, wherein said relay comprises a RTP relay.

43. The server of claim 32, wherein the server is separated from the plurality of relays.