Voice virtual private network

• US 9,979,828 B1
• Filed: 05/07/2014
• Issued: 05/22/2018
• Est. Priority Date: 04/12/2005
• Status: Active Grant

• Current Assignee: ZAMA Innovations LLC
• Original Assignee: TP Lab, Inc.
• Inventors: Chiu, Shin Cheung Simon, Ho, Chi Fai
• Primary Examiner(s): Shah, Chirag
• Assistant Examiner(s): Smith, Joshua
• Application Number:
  US14/272,372
• Time in Patent Office: 1,476 Days
• CPC Class Codes: H04L 12/4641 Virtual LANs, VLANs, e.g. virtual private networks [VPN] LAN interconnection over a bridge based backbone H04L12/462; encapsulation techniques H04L12/4633; routing of packets H04L45/00; packet switches H04L49/00; virtual private networks for security H04L63/0272
• CPC Class Codes: H04L 65/1046 Call controllers; Call servers
• CPC Class Codes: H04L 65/1056 Multi-site
• CPC Class Codes: H04M 2203/1091 Fixed mobile conversion
• CPC Class Codes: H04M 3/42221 Conversation recording systems at the subscriber's set H04M1/656
• CPC Class Codes: H04M 3/4234 Remote access to features of PBX or home telephone systems-teleworking in a PBX
• CPC Class Codes: H04M 3/56 Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities video conference systems H04N7/15
• CPC Class Codes: H04W 4/16 Communication-related supplementary services, e.g. call-transfer or call-hold

Abstract

Embodiments of the present invention include a Voice Virtual Private Network (Voice VPN) and methods for providing access to remote peer users across same. In one embodiment, voice and communication services of one or more corporate network are provided to remote user terminals across a Voice VPN Gateway. The level and type of access to voice and communication services provided can be determined statically, dynamically, or adaptively based on user data or user habits or history.

Claims

1. A system for providing teleconferencing services, comprising:

a server within a private data network for providing teleconferencing services within the private data network;

a user terminal external to the private data network and directly coupled to a public network, the user terminal comprising a session initiation protocol (SIP) user agent; and

a SIP-based teleconferencing gateway coupled to the server via the private data network and coupled to the user terminal via the public network, wherein the SIP-based teleconferencing gateway;

receives a SIP registration request from the SIP user agent of the user terminal to receive teleconferencing services within the private data network,

in response to the SIP registration request, retrieves user terminal information for receiving the teleconferencing services from a data store of the SIP-based teleconferencing gateway, and

sends the retrieved user terminal information to the server to activate the user terminal for receiving the teleconferencing services;

wherein the server;

receives the user terminal information from the SIP-based teleconferencing gateway,

activates the user terminal for teleconferencing services using the user terminal information, and

after activating the user terminal, sends to the SIP-based teleconferencing gateway a request for a teleconference call to the user terminal as a callee of the teleconference call, wherein the request comprises a callee telephone number, wherein the teleconference call comprises an Internet protocol (IP)-based voice call; and

wherein the SIP-based teleconferencing gateway further;

receives the request for the teleconference call from the server,

compares the callee telephone number in the request with a plurality of user terminals activated for the teleconferencing services within the private data network;

determines that the callee telephone number in the request matches the user terminal;

sends a SIP notification to the user terminal of the request for the teleconference call to the user terminal,

receives an acceptance of the teleconference call from the user terminal,

after receiving the acceptance of the teleconference call from the user terminal;

reserves resources for the IP-based voice call of the teleconference call,

establishes with the server a server side IP-based voice session associated with the IP-based voice call using the resources reserved for the IP-based voice call,

establishes with the user terminal a user side SIP-based voice session associated with the IP-based voice call using the resources reserved for the IP-based voice call, and

mediates audio signals of the IP-based voice call between the user side SIP-based voice session and the server side IP-based voice session using the resources reserved for the IP-based voice call, wherein the SIP-based teleconferencing gateway comprises an interface,

wherein in mediating the audio signals of the IP-based voice call, the interface transmits the audio signals between the user side SIP-based voice session and the server side IP-based voice session.

2. The system of claim 1, wherein the user terminal detects inactivity of the user side voice session.

3. The system of claim 2, wherein the user terminal terminates the user side voice session upon detecting inactivity of the user side voice session.

4. The system of claim 1, wherein the SIP-based teleconferencing gateway receives the SIP registration request using a web-based technology.

5. The system of claim 4, wherein the web-based technology comprises one or more of the following:

a web service;

hypertext transfer protocol (HTTP); and

Java.

6. The system of claim 1, wherein the public network comprises one or more of the following:

a cellular data network;

a wireless network; and

Internet.

7. The system of claim 1, wherein the SIP-based teleconferencing gateway encrypts signals for the user side voice session.

8. The system of claim 1, wherein the SIP-based teleconferencing gateway establishes a signaling session with the user terminal.

9. The system of claim 8, wherein the SIP-based teleconferencing gateway receives a recording request from the user terminal over the signaling session, wherein the recording request is to record the teleconferencing call.

10. The system of claim 1, wherein in mediating the audio signals of the IP-based voice call, the interface:

samples and digitizes the audio signals of the IP-based voice call;

places the sampled and digitized audio signals of the IP-based voice call into one or more voice packets; and

transmits the one or more the voice packets between the user side SIP-based voice session and the server side IP-based voice session.

11. The system of claim 1, wherein the server comprises a SIP server agent, the user terminal comprises the SIP user agent, and the SIP-based teleconferencing gateway comprises a SIP proxy, wherein the server and the SIP-based teleconferencing gateway communicate through the SIP server agent and the SIP proxy, wherein the user terminal and the SIP-based teleconferencing gateway communicate through the SIP user agent and the SIP proxy.

12. A method for providing teleconferencing service, comprising:

(a) receiving, by a session initiation protocol (SIP)-based teleconferencing gateway coupled to a server within the private data network, a SIP registration request from a SIP user agent of a user terminal to receive teleconferencing services within the private data network, the user terminal being external to the private data network and directly coupled to a public network, the SIP-based teleconferencing gateway being coupled to the user terminal via the public network;

(b) in response to the SIP registration request, retrieving user terminal information for receiving the teleconferencing services from a data store of the SIP-based teleconferencing gateway;

(c) sending, by the SIP-based teleconferencing gateway, the retrieved user terminal information to the server to activate the user terminal for receiving teleconferencing services;

(d) receiving, by the server, the user terminal information from the SIP-based teleconferencing gateway;

(e) activating, by the server, the user terminal for teleconferencing services using the user terminal information;

(f) after activating the user terminal, sending, by the server to the SIP-based teleconferencing gateway, a request for a teleconference call to the user terminal as a callee of the teleconference call, wherein the request comprises a callee telephone number, wherein the teleconference call comprises an Internet protocol (IP)-based voice call;

(g) receiving, by the SIP-based teleconferencing gateway, the request for the teleconference call from the server;

(h) comparing, by the SIP-based teleconferencing gateway, the callee telephone number in the request with a plurality of user terminals activated for the teleconferencing services within the private data network;

(i) determining, by the SIP-based teleconferencing gateway, that the callee telephone number in the request matches the user terminal;

(j) sending, by the SIP-based teleconferencing gateway, a SIP notification to the user terminal of the request for the teleconference call to the user terminal;

(k) receiving, by the SIP-based teleconferencing gateway, an acceptance of the teleconference call from the user terminal;

(l) after receiving the acceptance of the teleconference call from the user terminal;

(1) reserving, by the SIP-based teleconferencing gateway, resources for the IP-based voice call of the teleconference call;

(2) establishing, by the SIP-based teleconferencing gateway with the server, a server side IP-based voice session associated with the IP-based voice call using the resources reserved for the IP-based voice call;

(3) establishing, by the SIP-based teleconferencing gateway with the user terminal, a user side SIP-based voice session associated with the IP-based voice call using the resources reserved for the IP-based voice call; and

(4) mediating, by the SIP-based teleconferencing gateway, audio signals of the IP-based voice call between the user side SIP-based voice session and the server side IP-based voice session using the resources reserved for the IP-based voice call, wherein the SIP-based teleconferencing gateway comprises an interface, wherein the mediating comprises;

transmitting, by the interface, the audio signals between the user side SIP-based voice session and the server side IP-based voice session.

13. The method of claim 12, further comprising:

(m) detecting, by the user terminal, inactivity of the user side voice session.

14. The method of claim 13, further comprising:

(n) terminating, by the user terminal, the user side SIP-based voice session upon detecting inactivity of the user side SIP-based voice session.

15. The method of claim 12, wherein the receiving (a) comprises:

(a1) receiving, by the SIP-based teleconferencing gateway, the SIP registration request using a web-based technology, wherein the web-based technology comprises one or more of the following;

a web service;

hypertext transfer protocol (HTTP); and

Java.

16. The method of claim 12, wherein the public network comprises one or more of the following:

a cellular data network;

a wireless network; and

Internet.

17. The method of claim 12, wherein the establishing (l3) comprises:

(l3i) establishing, by the SIP-based teleconferencing gateway, a signaling session with the user terminal, wherein the method further comprises;

(m) receiving, by the SIP-based teleconferencing gateway, a recording request from the user terminal over the signaling session, wherein the recording request is to record the teleconference call.

18. The method of claim 12, wherein the mediating comprises:

sampling and digitizing the audio signals of the IP-based voice call by the interface;

placing the sampled and digitized audio signals of the IP-based voice call into one or more voice packets by the interface; and

transmitting, by the interface, the one or more the voice packets between the user side SIP-based voice session and the server side IP-based voice session.

19. The method of claim 12, wherein the server comprises a SIP server agent, the user terminal comprises the SIP user agent, and the SIP-based teleconferencing gateway comprises a SIP proxy, wherein the server and the SIP-based teleconferencing gateway communicate through the SIP server agent and the SIP proxy, wherein the user terminal and the SIP-based teleconferencing gateway communicate through the SIP user agent and the SIP proxy.