Patents by Inventor Madhusudhan R. Adupala

Madhusudhan R. Adupala has filed for patents to protect the following inventions. This listing includes patent applications that are pending as well as patents that have already been granted by the United States Patent and Trademark Office (USPTO).

  • Patent number: 11627171
    Abstract: Systems and techniques are provided for voice calling with a connected device that does not include a SIM card or telephone port. Outgoing audio data may be received at an embedded browser running on a connected device, may be sent using Web Real Time Communications (WebRTC) from the embedded browser to an integration layer running within the embedded browser, and may be sent from the integration layer to a border controller for a voice call carrier over a Session Initiation Protocol (SIP) connection according to Secure Real Time Transport Protocol (SRTP). Incoming audio data may be received at the integration layer from the border controller for the voice call carrier over the SIP connection according to SRTP, may be sent using WebRTC from the integration layer to the embedded browser, and may be sent from the embedded browser to an audio output of the connected device which may output audio.
    Type: Grant
    Filed: November 22, 2021
    Date of Patent: April 11, 2023
    Assignee: Google LLC
    Inventors: Jeffrey Ching Wang, Chien-Jung Kung, Madhusudhan R. Adupala
  • Publication number: 20220094720
    Abstract: Systems and techniques are provided for voice calling with a connected device that does not include a SIM card or telephone port. Outgoing audio data may be received at an embedded browser running on a connected device, may be sent using Web Real Time Communications (WebRTC) from the embedded browser to an integration layer running within the embedded browser, and may be sent from the integration layer to a border controller for a voice call carrier over a Session Initiation Protocol (SIP) connection according to Secure Real Time Transport Protocol (SRTP). Incoming audio data may be received at the integration layer from the border controller for the voice call carrier over the SIP connection according to SRTP, may be sent using WebRTC from the integration layer to the embedded browser, and may be sent from the embedded browser to an audio output of the connected device which may output audio.
    Type: Application
    Filed: November 22, 2021
    Publication date: March 24, 2022
    Inventors: Jeffrey Ching Wang, Chien-Jung Kung, Madhusudhan R. Adupala
  • Patent number: 11184408
    Abstract: Systems and techniques are provided for voice calling with a connected device that does not include a SIM card or telephone port. Outgoing audio data may be received at an embedded browser running on a connected device, may be sent using Web Real Time Communications (WebRTC) from the embedded browser to an integration layer panning within the embedded browser, and may be sent from the integration layer to a border controller for a voice call carrier over a Session Initiation Protocol (SIP) connection according to Secure Real Time Transport Protocol (SRTP). Incoming audio data may be received at the integration layer from the border controller for the voice call carrier over the SIP connection according to SRTP, may be sent using WebRTC from the integration layer to the embedded browser, and may be sent from the embedded browser to an audio output of the connected device which may output audio.
    Type: Grant
    Filed: February 12, 2020
    Date of Patent: November 23, 2021
    Assignee: GOOGLE LLC
    Inventors: Jeffrey Ching Wang, Chien-Jung Kung, Madhusudhan R. Adupala
  • Publication number: 20210227000
    Abstract: Systems and techniques are provided for voice calling with a connected device that does not include a SIM card or telephone port. Outgoing audio data may be received at an embedded browser running on a connected device, may be sent using Web Real Time Communications (WebRTC) from the embedded browser to an integration layer panning within the embedded browser, and may be sent from the integration layer to a border controller for a voice call carrier over a Session Initiation Protocol (SIP) connection according to Secure Real Time Transport Protocol (SRTP). Incoming audio data may be received at the integration layer from the border controller for the voice call carrier over the SIP connection according to SRTP, may be sent using WebRTC from the integration layer to the embedded browser, and may be sent from the embedded browser to an audio output of the connected device which may output audio.
    Type: Application
    Filed: February 12, 2020
    Publication date: July 22, 2021
    Inventors: Jeffrey Ching Wang, Chien-Jung Kung, Madhusudhan R. Adupala