Dynamic VoIP routing and adjustment
Systems and processes are provided for dynamically adjusting and routing signals and communications. According to one aspect, one or more codecs applied to the signals may be switched or changed to manage signal quality. According to another aspect, communications may be re-routed to an intended recipient entirely over the Internet. The systems and processes described herein are especially well suited for use in connection with detention environments.
1. A communications system for a detention environment, said system comprising:
terminals for generating and receiving audio signals, said terminals being located within the detention environment, for use by detained residents of the detention environment;
a monitoring device that provides continuous monitoring throughout the duration of a communication of voice biometrics verification and quality of voice;
an evaluation device for evaluating communications signals corresponding to the audio signal, and generating an output signal upon detection of an event from the communication signal, the evaluation device being capable of detecting each of the following events: (1) speech of a predetermined language, (2) a participant being flagged as a subject of interest that is detected by voice biometrics verification, and (3) use of a video conferencing system or a peer-to-peer audio system; and
a gateway device for receiving the output signal from the evaluation device, and for changing one or more codecs applied to the audio signals, such that the gateway device changes the one or more codecs to a higher quality codec in response to the event indicated by the output signal.
2. The system of claim 1 , wherein the gateway device is located within one of the terminals.
3. The system of claim 1 , wherein the quality of the voice in the communications signals is measured according to a reference value.
4. The system of claim 3 , wherein the gateway device is responsive to a determination that the measured quality of voice is outside of a predetermined quality range.
5. The system of claim 1 , wherein the terminals are analog telephone terminals and the gateway device comprises an analog telephone adapter for converting an analog voice signal to a digital voice signal.
6. The system of claim 1 , wherein the terminals are VOIP-enabled devices, and the terminals convert analog voice signals to VoIP codecs.
7. The system of claim 1 , further comprising a first server for caching a version of the audio signals corresponding to a maximum available bandwidth and a second server, wherein the first server uploads the version of the audio signals corresponding to maximum available bandwidth to the second server.
8. The system of claim 1 , the event further comprising the determination that speech or speaker recognition is desired.
9. The system of claim 8 , wherein the one or more codecs are changed to a high-quality voice codec when the event is a participant being flagged as a subject of interest.
10. The system of claim 1 , wherein the gateway device is capable of routing a call away from a public switched telephone network (PSTN) when communicating with a VoIP-enabled device.
11. A communications system for a detention environment, said system comprising:
terminals for generating and receiving audio signals, said terminals being located within the detention environment, for use by detained residents of the detention environment;
a monitoring device that provides continuous monitoring throughout the duration of a communication of voice biometrics verification and quality of voice;
an evaluation device for evaluating communications signals corresponding to the audio signal, and for generating an output signal based on the evaluation, wherein the output signal is indicative of at least an event indicated by the communications signals; and
a gateway device for receiving the output signal from the evaluation device, and for changing one or more codecs applied to the audio signals, such that the gateway device changes the one or more codecs to a higher quality codec in response to the event indicated by the output signal,
wherein the gateway device is capable of routing a call away from a public switched telephone network (PSTN) when communicating with a VoIP-enabled device.