Abstract: A method of making Voice over Internet Protocol (VoIP) calls, legacy circuit calls, sending/receiving Short Message Service (SMS) over Long Term Evolution (LTE) modem or a legacy modem on a mobile terminal with both kinds of modems, and providing all legacy modem functions is disclosed. In addition, methods for dynamic selection of radio in a mobile terminal capable of Rich Communications Services (RCS) capabilities, and a method for redirecting RCS traffic to an alternate network interface is also disclosed. Methods for Session Initiation Protocol module (SIP) stack functions to be distributed across different processors on a mobile terminal, and directed to different network interfaces is disclosed. Methods of adding video calling and RCS functions without encountering the dual registration problem are also disclosed.

Abstract: Methods of IP to IP handover for VoIP calls are disclosed. Handover can occur using UDP in connectionless mode to switch networks, a hybrid approach including establishing both a TCP session and a UDP session and handing off using the UDP session, utilizing a network SIP proxy to maintain connection with the IMS server while the mobile device switches networks connecting the mobile device to the SIP proxy, using a modified Connection Manager that permits multiple IP connections simultaneously, and connecting via an on-device virtual network to an on-device router which reroutes calls to the appropriate network are detailed.

Abstract: A method of making Voice over Internet Protocol (VoIP) calls, legacy circuit calls, sending/receiving Short Message Service (SMS) over Long Term Evolution (LTE) modem or a legacy modem on a mobile terminal with both kinds of modems, and providing all legacy modem functions is disclosed. In addition, methods for dynamic selection of radio in a mobile terminal capable of Rich Communications Services (RCS) capabilities, and a method for redirecting RCS traffic to an alternate network interface is also disclosed. Methods for Session Initiation Protocol module (SIP) stack functions to be distributed across different processors on a mobile terminal, and directed to different network interfaces is disclosed. Methods of adding video calling and RCS functions without encountering the dual registration problem are also disclosed.

Abstract: A method and system for utilizing information content in speech and a transition hangover between speech and noise to generate comfort noise on the decoder side. This adaptation to noise may be accomplished using various algorithms of estimating the spectrum of color noise. According to an embodiment of the present invention, an adaptation algorithm may be implemented that adapts, with time, rather than a block based algorithm to prevent the repeated generation of artifacts present in the block that are being adapting to. The method and system of the present invention generates comfort noise in the absence of silent insertion descriptions containing spectrum information thereby saving bandwidth and generating colored comfort noise that reflects the spectrum of the actual noise.

Abstract: The present invention relates to an algorithm for ensuring compliancy of an algorithm module when integrated in a real time software system. The compliancy tests may include a memory test, interrupt test, latency test and other tests, as well as combinations thereof. An inventive aspect of the present invention relates to a unit test harness for verifying that a software algorithm module meets performance and functional requirements when integrated in a complete real-time software system. A software algorithm module eliminates or reduces unwanted behavior by the caller or other software on a real-time software system due to incorrect operations, which may involve interrupts, memory usage, register usage and/or other factors.

Abstract: A method and system for implementing a simple Gaussian white noise generator for real time speech synthesis applications. According to an embodiment of the present invention, a method for generating a Gaussian random noise signal comprises the steps of: generating a first predetermined number of outputs from a first random number generator having a first period; generating a second predetermined number of outputs from a second random number generator having a second period different from the first period; calculating a first average from a first combination of inputs from the first random number generator and the second random number generator; calculating a second average from a second combination of inputs from the first random number generator and the second random number generator; and generating a Gaussian output based on the first average and the second average.

Abstract: A method and system for utilizing information content in speech and a transition hangover between speech and noise to generate comfort noise on the decoder side. This adaptation to noise may be accomplished using various algorithms of estimating the spectrum of color noise. According to an embodiment of the present invention, an adaptation algorithm may be implemented that adapts with time, rather than a block based algorithm to prevent the repeated generation of artifacts present in the block that are being adapting to. The method and system of the present invention generates comfort noise in the absence of silent insertion descriptions containing spectrum information thereby saving bandwidth and generating colored comfort noise that reflects the spectrum of the actual noise.

Abstract: The present invention is directed to a voice tone discriminator for distinguishing between call progress tones and voice. The voice tone discriminator is useful in various applications involving the ability to automatically charge a telephone user based on the exact time the user starts speaking. According to another aspect of the present invention, a prediction algorithm may be implemented for distinguishing voice and tone based on the fact that tones are more accurately modeled with a linear filter than voice signals. Thus, a low order filter or predictor may accurately model redundancies in tones but not in voice. A normalized error between an original and a predicted signal may be used to distinguish voice from tones. Voice may be detected when the error is above a preset threshold for a time greater than a preset fixed duration.