SYSTEM AND METHOD FOR ESTABLISHING UNIVERSAL REAL TIME PROTOCOL BRIDGING

• US 20070171898A1
• Filed: 11/28/2006
• Published: 07/26/2007
• Est. Priority Date: 11/29/2005
• Status: Abandoned Application

• Current Assignee: Xteltek LLC
• Original Assignee: Xteltek LLC
• Inventors: Salva, Paul
• Application Number:
  US11/563,705
• Publication Number:
  US 20070171898A1
• Time in Patent Office: Days
• US Class Current: 370/356
• CPC Class Codes: H04L 63/08 for authentication of entities cryptographic mechanisms or cryptographic arrangements for entity authentication H04L9/32
• CPC Class Codes: H04L 65/1026 at the edge
• CPC Class Codes: H04L 65/1033 Signalling gateways
• CPC Class Codes: H04L 65/1036 at the edge
• CPC Class Codes: H04L 65/1101 Session protocols
• CPC Class Codes: H04L 65/1104 Session initiation protocol [SIP]
• CPC Class Codes: H04L 65/403 Arrangements for multi-party communication, e.g. for conferences data switching systems for conference H04L12/18; arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities H04M3/56; television conferencing systems H04N7/15
• CPC Class Codes: H04L 65/65 Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
• CPC Class Codes: H04M 2203/5018 Initiating a conference during a two-party conversation, i.e. three-party service or three-way call
• 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

Abstract

An apparatus and methods for Real Time Protocol (RTP) based bridging to create calls to and From any communication devices connected to the Public Switched Telephone Network (PSTN) or any public or private Internet Protocol network. The methods are represented in controls residing either in an RTP Bridge apparatus or as middleware on a service creation platform, softswitch, or SIP proxy services incorporating SIP bridging and SIP call relay technologies, with an interface to a TDM (Time Division Multiplexing) switch operated by a Local Exchange Carrier (LEC) or Competitive LEC (CLEC), or hosted by an Internet Telephony Service Provider (ITSP) with connectivity to the PSTN through media gateways. The method permits a party to request an on demand conference call by either dialing into the apparatus from the PSTN or from any type of RTP communication device such as an IP phone; or using a form of signaling from RTP translation device. The moderator initiates the request, enters the participant(s) to be included in the call, and the launch sequence is initiated and the apparatus makes contact with the phone/end point and uses RTP bridging to interlace the packet streams and deliver one stream back to all the devices. The advantage is an on demand conference call to one or many participants, on similar or different communications platforms.

Claims

1. A telecommunications apparatus using RTP based bridging for establishing interlaced, two-way audio communications between communications devices connected to both PSTN and IP communications networks, the telecommunications apparatus comprising:

means for receiving data from at least one of a calling parties TDM phone, a calling, party'"'"'s intelligent communications device, and a calling parties alternative SIP signaling device;

means for authenticating the calling party'"'"'s identify using at least one of ANI, Caller ID, PIN number, IVR, IP address, or a signaling protocol;

means for recording the calling party'"'"'s identity for later identification;

means for requesting the calling party to identify at least one party to be called by entry of a number or address associated with called party'"'"'s telecommunications equipment;

means for placing at least one call to a local PSTN telephone number using at least one of SIP-PSTN proxy services, IP-PSTN, direct connection to a SIP telephone number, or sending RTP packets to a communications device capable of recording or converting RTP data streams to a different media type.

2. The apparatus according to claim 1, wherein the means for requesting the calling party to identify at least one party to be called by entry of a number or address associated with called party'"'"'s telecommunications equipment includes means for receiving conference call launch data, comprising at least one of:

means for receiving and translating touch tone signals using Interactive Voice Response (IVR);

means for receiving HTTP data from a web browser;

means for receiving WAP data from a WAP-enabled device; and

means for receiving signaling data from a VoIP-capable device.

3. The apparatus as claimed in claim 1, wherein the RTP bridging, authentication, and pre-programmed launch sequence instructions are done by the same machine such as an appliance.

4. The apparatus according to claim 1 wherein the apparatus comprises a software application residing within at least one of a SIP Bridge, a softswitch an SIP Proxy Server, or a service creation platform separate from a high density telco switch or Media Gateway port.

5. A method for establishing a conference call between a calling party and a called party using an RTP bridge device, the method comprising the steps of:

from a calling device of the calling party, calling a predefined telephone number that terminates to the RTP bridge device;

recording an identifier of the calling device, the identifier comprising at least one of an ANI, SIP number, or IP addresser other signaling device;

authenticating the identifier against a registration database;

dropping the call received from the calling device;

dialing back the calling device;

providing instructions to the calling party via the calling device, prompting the calling party to enter at least one number corresponding to a called device of the called party in a conference call;

determining a type of communications protocol used by the called device; and

establishing a native form connection to the called device by translating the communications protocol using at least one of software codecs and hardware micro controller units into a common RTP data stream; and

passing through the identifier of the calling device to the called device as an indication to the called party to recognize an incoming request for communications.

6. The method according to claim 5, wherein the calling party acts as a first moderator for the conference call, further comprising the steps of:

dialing back a predetermined, alternative number or address corresponding to a second moderator; and

permitting the second moderator to perform at least one of terminating the conference call, listening in on the conference call, and recording the conference call.

7. The method according to claim 5, wherein the calling device is a WAP-enabled device, further comprising the step of sending the RTP bridging device a set of preprogrammed instructions for the conference call, the preprogrammed instructions including a time limit for a duration of the conference call entered using a GUI interface provided to the WAP-enabled device.

8. The method according to claim 5, further comprising the step of signaling the RTP Bridge device to executer a preprogrammed launch sequence to initiate the conference call.

9. The method according to claim 5, further comprising the step of recording at least one of audio and video data associated with the conference call.

10. The method according to claim 9, further comprising the step of duplicating the at least one of audio and video data for broadcast or multicast distribution.

11. The method according to claim 9 further comprising the step of playing backs the recorded data.

12. The method according to claim 5, further comprising the step of signaling the RTP Bridge device to capture and stream on demand, do as to simulate a push-to-talk or walkie talkie operation from any communications device participating in the conference call.

13. The method according to claim 5, wherein the step of calling the predefined telephone number that terminates to the RTP bridge device comprises using a Ring Down Circuit to signal the RTP bridge device, thereby permitting the RTP bridge device to drop the call and initiate a call back to the calling party.

14. The method according to claim 5, further comprising the step of creating a peer-to-peer connection between the calling and called devices.

15. The method according to claim 14 wherein the peer-to-peer connection uses a multicast tunnel.

16. The method according to claim 14, wherein the peer-to-peer connection uses data encryption.

17. The method according to claim 14, wherein the peer-to-peer connection carries at least one of private voice, video, or data communications.

18. The method according to claim 5, further comprising the step of delivering advertising information to at least one of the calling party and the called party.

19. The method according to claim 18, wherein the advertising information comprises audio information delivered from a server associated with the RTP bridge device.

20. The method according to claim 19, wherein the audio information is interactive audio information.