System, method, and computer program product for connecting or coupling audio communications systems over a software defined wide area network

• US 10,484,513 B2
• Filed: 04/07/2018
• Issued: 11/19/2019
• Est. Priority Date: 07/17/2015
• Status: Active Grant

• Current Assignee: Nsgdatacom Incorporated
• Original Assignee: Nsgdatacom Incorporated
• Inventors: King, Graham, Tiley, Simon
• Primary Examiner(s): Yeung, Mang Hang
• Application Number:
  US15/947,788
• Publication Number:
  US 20180227394A1
• Time in Patent Office: 591 Days
• Field of Search: 370328
• CPC Class Codes: G10L 19/02 using spectral analysis, e.g. transform vocoders or subband vocoders
• CPC Class Codes: G10L 19/167 Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
• CPC Class Codes: H04L 67/025 for remote control or remote monitoring of applications
• CPC Class Codes: H04L 67/566 Grouping or aggregating service requests, e.g. for unified processing
• CPC Class Codes: H04L 67/59 Providing operational support to end devices by off-loading in the network or by emulation, e.g. when they are unavailable
• CPC Class Codes: H04L 69/04 Protocols for data compression, e.g. ROHC
• CPC Class Codes: H04L 69/08 Protocols for interworking; Protocol conversion
• CPC Class Codes: H04L 69/085 specially adapted for interworking of IP-based networks with other networks
• CPC Class Codes: H04L 69/324 in the data link layer [OSI layer 2], e.g. HDLC

Abstract

An automated telecommunications system includes a first system that receives audio frequency signals such as PSTN-compatible, voice over IP, or high definition voice, decodes and interprets said incoming signals according to the type and format, and transmits digital messages to a second system over a software defined wide area network (SDWAN). Said second system receives and interprets digital messages incoming from the first system, encodes and regenerates outgoing audio frequency signals. The system may be bi-directional and operate over a software defined wide area network, such as an IP based wireless or wired data network. The functionality of said first and second systems may be combined at a single location and interface with a conventional PSTN compatible VoIP system or High Definition Voice system at either or both ends and allow PSTN, conventional VoIP and High Definition Voice communications to share the same packet data stream over a SDWAN connection.

Claims

1. A system of transmitting and receiving audio frequency signals comprising PSTN-compatible, or voice over IP (VoIP) signals, or high definition voice signals over a packet data network wherein the packet data network comprises a software defined wide area network (SDWAN), the system comprising:

a first system comprising;

at least one first memory device; and

at least one first processor, coupled to said at least one first memory device and the packet data network, said at least one first processor comprising at least one or more of;

at least one application-specific integrated circuit (ASIC),

at least one chipset,

at least one processor,

at least one microprocessor,

at least one logic circuit,

at least one quantum computer,

at least one multi-core processor,

at least one quad core processor,

at least one programmable logic device (PLD),

at least one field programmable gate arrays (FPGA),

at least one information appliance,

at least one system on a chip (SOC),

at least one multi-processor,

at least one computer processor,

at least one graphics processor, or

at least one digital signal processor (DSP),

said at least one first processor configured to;

receive one or more first incoming audio frequency signals;

digitize said one or more first incoming audio frequency signals according to an audio standard to obtain first digitized audio frequency signals;

segment said first digitized audio frequency signals to generate one or more first sequences of audio signal samples, each of said one or more first sequences of audio signal samples comprising a first preprogrammed sample size;

compress said each of said one or more first sequences of audio signal samples according to a first preprogrammed set of rules comprising a first preprogrammed compression algorithm, to produce a first sequence of one or more strings of processed samples, responsive to determining that said each of said one or more first sequences of audio signal samples is determined to meet a first preprogrammed criteria for compression according to said first preprogrammed set of rules;

accumulate said first sequence of said one or more strings of processed samples to create a first group of samples ready for transmission according to a first aggregation threshold defined by a second preprogrammed set of rules,

create a first outgoing digital message from said first group of samples ready for transmission according to a forwarding threshold defined by a third preprogrammed set of rules, using at least one of;

a first pre-defined data link protocol, or

a first control channel;

transmit said first outgoing digital message over the software defined wide area network (SDWAN) to a second system;

receive, from the software defined wide area network, and interpret, a first incoming digital message from the second system;

process said first incoming digital message into one or more incoming digital audio stream samples according to said third pre-programmed set of rules comprising said first pre-defined data link protocol, or said first control channel, or a second pre-defined data link protocol, or a second control channel; and

accumulate and process said one or more incoming digital audio stream samples according to a fourth preprogrammed set of rules comprising being configured to;

decompress any compressed of said one or more incoming digital audio stream samples according to a first preprogrammed decompression algorithm defined by said fourth preprogrammed set of rules into decompressed digital audio samples;

accumulate said first decompressed digital samples into a first buffer according to a first jitter buffer threshold, wherein said first jitter buffer threshold is defined by said fourth preprogrammed set of rules; and

regenerate first outgoing audio frequency signals based on said first decompressed digital audio samples.

2. The system of claim 1, further comprising at least one or more of:

wherein the software defined wide area network comprises an Internet protocol (IP) based network;

or

wherein the software defined wide area network comprises an LTE software defined wide area network;

or

an interface to a voice over Internet protocol (VoIP) packet system or a High Definition Voice system.

3. The system of claim 1, further comprising:

the second system coupled to said first system, said second system comprising;

at least one second memory device; and

at least one second processor, coupled to said at least one second memory device, said at least one second processor comprising at least one or more of;

at least one application-specific integrated circuit (ASIC),

at least one chipset,

at least one processor,

at least one microprocessor,

at least one logic circuit,

at least one quantum computer,

at least one multi-core processor,

at least one quad core processor,

at least one programmable logic device (PLD),

at least one field programmable gate arrays (FPGA),

at least one information appliance,

at least one system on a chip (SOC),

at least one multi-processor,

at least one computer processor,

at least one graphics processor, or

at least one digital signal processor,

said at least one second processor configured to

receive one or more second incoming audio frequency signals;

digitize said one or more second incoming audio frequency signals according to an audio standard to obtain second digitized audio frequency signals;

segment said second digitized audio frequency signals to generate one or more second sequences of audio signal samples, each of said one or more second sequences of said audio signal samples comprising said first preprogrammed sample size or a second preprogrammed sample size;

compress said each of said one or more second sequences of audio signal samples according to said first preprogrammed set of rules or a fifth preprogrammed set of rules comprising said first preprogrammed compression algorithm or a second preprogrammed compression algorithm, to produce a second sequence of one or more strings of processed samples, responsive to determining that said each of said one or more second sequences of audio signal samples is determined to meet said first preprogrammed criteria, or a second preprogrammed criteria for compression according to said first or said fifth preprogrammed set of rules;

accumulate said second sequence of said one or more strings of processed samples to create a second group of samples ready for transmission according to said first aggregation threshold, or a second aggregation threshold defined by a sixth preprogrammed set of rules;

create a second outgoing digital message from said second group of samples ready for transmission according to said forwarding threshold, or a second forwarding threshold defined by a seventh preprogrammed set of rules, wherein said second outgoing digital message for transmission is created using said first or said second pre-defined data link protocol, or said first or said second control channel, or a third pre-defined data link protocol, or a third control channel;

transmit said second outgoing digital message over the software defined wide area network to said first system;

receive at the second system from the software defined wide area network, and interpret, a second incoming digital message from said first system; and

process said second incoming digital message into one or more second incoming digital audio stream samples according to said third preprogrammed set of rules or an eighth preprogrammed set of rules comprising said first, said second, or said third pre-defined data link protocol, or said first, said second, or said third control channel, or a fourth pre-defined data link protocol or a fourth control channel;

accumulate and process said one or more second incoming digital audio stream samples according to said fourth preprogrammed set of rules, or a ninth preprogrammed set of rules;

decompress said one or more of said second incoming digital audio sample streams according to a second pre-programmed decompression algorithm as defined by said fourth or said ninth pre-programmed set of rules into second decompressed digital audio samples;

accumulate said second decompressed digital audio samples into a second buffer according to a second jitter buffer threshold, wherein said second jitter buffer threshold is defined by said fourth or said ninth preprogrammed set of rules; and

regenerate second outgoing audio frequency signals based on said second decompressed digital audio samples.

4. The system of claim 3, wherein said first system and the second system are located at least one of:

at a single location, or

at different locations.

5. The system of claim 3, wherein the system is configured to use at least one of:

predetermined information, or

learned information, or

preconfigured information,

to determine at least one of said first, said second, said third, or said fourth preprogrammed rules to apply to process said first or said second signal information and forward messages between said first and the second systems.

6. The system of claim 3, wherein said ninth preprogrammed set of rules comprises configuring the system to at least one of any one or more of:

define a size of said second jitter buffer;

or

define a size of said second jitter buffer on a call by call basis;

or

define a size of said jitter buffer based on at least one of any one or more of;

a predetermined criteria, or

a fixed criteria, or

dynamically collected operational data,

wherein said dynamically collected operational data comprises at least one or more of;

current network jitter, or

current network delay, or

current network loading, or

packet loss, or

network data errors, or

time of day, or

QoS requirements, or

other relevant collected data

wherein said dynamically collected operational data comprises being

configured to be at least one of any one or more of;

collected locally, or

sourced from externally connected equipment comprising at least

one of any one or more of;

a router, or

a SDWAN device, or

a SDWAN controller, or

performance monitoring equipment.

7. The system of claim 1, wherein the system further comprises wherein at least one of:

said first system, or the second system, is configured to at least one of;

compress, or

decompress, at least one of;

PSTN, or

VoIP, or

High Definition Voice,

compatible audio signals.

8. The system of claim 1, comprising:

wherein said first control channel comprises at least one of;

an in-band control channel, or

an out-of-band control channel, and

wherein said first control channel is configured to;

remotely manage and provide a mechanism to pass control information for synchronization between, said first and the second systems,

wherein said mechanism is configured to pass said control information at least one of;

during initialization, or

during routing changes, or

to enable changes in size of jitter buffers, or

to turn compression on or off during a call, and

provide communications to perform at least one of;

facilitate transportation of a plurality of strings of said processed samples with a single string of digital messages;

or

provide monitoring function;

or

provide a control function;

or

determine real time diagnostic information;

or

determine status information;

or

determine ancillary information.

9. The system according to claim 1, wherein said at least one first, second, third, or fourth preprogrammed set of rules comprises at least one of:

a) at least one intelligent, dynamic, or buffered preprogrammed rule based on positive feedback or collected information;

or

b) at least one preprogrammed rule dependent upon a type of the software defined wide area network over which the audio frequency signals are transmitted;

or

c) wherein said at least one of said at least one first, second, third, or fourth preprogrammed rule is configured to at least one of;

regenerate as accurately as possible, the audio frequency signals, or

reproduce a channel regardless of content transported, while said at least one first, second, third, or fourth preprogrammed rule maintains integrity of the audio frequency signals;

or

d) wherein determining whether voice over IP (VoIP) or High Definition Voice packets are received, either segmenting and compressing first, or responsive to determining that said packets are already compressed, then proceeding to accumulating audio signal samples, and combining and creating an aggregated digital message for transmitting;

or

e) wherein determining whether receiving and interpreting a second incoming digital message at the second system from said first system, processing to isolate a second incoming digital audio stream, and responsive to determining that the second incoming digital audio stream is going to a VoIP system or a High Definition Voice system, and then processing digital samples of said incoming digital message and outputting packets containing digital audio stream information, and responsive to determining that the second incoming digital audio stream is determined not to be going to said VoIP system or said High Definition Voice system, then first processing samples of the incoming digital messages including decompressing responsive to determining that decompression is required, accumulating processed samples in a jitter buffer, and regenerating a second outgoing audio signal;

or

f) setting an aggregation threshold and a forwarding threshold, receiving a third incoming digital audio signal, segmenting the third incoming digital audio signal into a segmented audio signal sample, compressing the segmented audio signal sample into a compressed audio signal sample, buffering in an aggregation buffer the compressed audio signal sample, and

determining whether the aggregation threshold is met, and responsive to the aggregation threshold being determined not met then

determining whether the incoming signal end has been reached and

responsive to determining that the incoming signal end is determined to be reached then ending, or

responsive to determining that the end of the incoming signal has not been reached then updating the aggregation threshold, and

responsive to determining that the aggregation threshold is determined to be met then

forwarding samples for transmission,

merging samples from multiple audio streams into a group of merged samples, and preparing for transmission of the group of merged samples, and then

determining whether the forwarding threshold is met, and

responsive to determining that the forwarding threshold is determined to be met, then

first setting a next forwarding threshold,

creating a digital message, and

transmitting the digital message from said first system to the second system over the software defined wide area network;

or

responsive to determining that the forwarding threshold is determined to not have been met then

proceeding immediately; and

updating the forwarding threshold.

10. The system according to claim 1,

wherein said first system configured to segment comprises being configured to:

reassemble and resegment said first digitized incoming audio frequency signals, responsive to determining that a sample size is different from a desired sample size.

11. The system according to claim 1,

wherein said first preprogrammed rules comprises being configured to:

compress, using said first preprogrammed compression algorithm, any of said each of said one or more first sequences of said audio signal samples responsive to determining that said first preprogrammed criteria for compression according to said first preprogrammed set of rules determines it is desired to compress said any of said each of said one or more first sequences of said audio signal samples; and

not compress any of said each of said one or more first sequences of audio signal samples, according to said first preprogrammed set of rules, responsive to determining that said first preprogrammed criteria for compression according to said first preprogrammed set of rules determines it is not desired to compress said any of said each of said one or more first sequences of said audio signal samples,

wherein said first preprogrammed criteria for compression according to said first preprogrammed set of rules comprises being configured to determine desirability of compression or noncompression is based on any one or more of;

responsive to determining that the audio is determined to have already been compressed;

or

fewer than a predefined number of calls are determined to be currently in progress;

or

responsive to determining that the call is from a wireless phone;

or

responsive to determining that the call is international;

or

responsive to determining that the call is between certain times of the day;

or

responsive to determining that the call is an emergency call;

or

responsive to determining that the call is an otherwise prioritized call;

or

responsive to determining that the call is a facsimile call;

or

responsive to determining that the call is a data modem call;

or

responsive to determining that the call meets other preprogrammed criteria;

or

responsive to determining that predetermined characteristics of the network are operating within certain performance limits comprising any one or more of the following;

network loading;

or

network delay;

or

network jitter;

or

network data errors;

or

packet loss;

or

responsive to determining that characteristics measured in real time of the network are operating within certain performance limits comprising any one or more of;

network loading;

or

network delay;

or

network jitter;

or

network data errors;

or

packet loss;

or

other relevant collected data.

12. The system according to claim 1,

wherein said second preprogrammed set of rules comprises being configured to:

define said first aggregation threshold that determines when a said first group of outgoing samples is ready for transmission; and

wherein said first aggregation threshold comprises being configured to be at least one of any one or more of;

fixed;

or

variable, based on a plurality of factors, wherein said plurality of factors comprises at least one of any one or more of;

a number of outgoing samples ready for transmission;

or

an amount of data that may be transmitted in a next packet;

or

a maximum packet size;

or

a timer;

or

a number of calls in progress;

or

a priority of certain call types;

or

other relevant operational data.

13. The system according to claim 1,

wherein said third preprogrammed set of rules comprises being configured to any one or more of:

merge to create one or more packets of data for transmission;

or

define an order of priority, responsive to determining that multiple packets are to be transmitted;

or

define frequency of when packets are to be sent;

or

define a time at which packets are to be sent;

or

define size of packets;

or

adjust the forwarding threshold based upon at least one of any one or more of;

taking into account real time factors comprising any one or more of;

current network operating characteristics comprising at least one of any one or more of;

loading;

or

delay;

or

jitter;

or

packet loss;

or

data errors;

or

availability of alternative routing;

or

availability of additional bandwidth on demand;

or

number of calls in progress;

or

time of day;

or

emergency network loading;

or

other network loading;

or

other relevant operational data;

or

Quality of Service (QoS) requirements;

or

Prioritization.

14. The system according to claim 1,

wherein said accumulate said first sequence of said one or more strings of processed samples to create said first group of samples ready for transmission according to said first aggregation threshold defined by said second preprogrammed set of rules, comprises wherein said first system comprises an aggregation buffer.

15. The system according to claim 1,

wherein said the software defined wide area network comprises a communications network comprising:

at least one wireless network;

or

at least one wired;

or

at least one satellite network;

or

at least one optical network.

16. The system according to claim 1, comprising:

wherein said first system configured to create comprises using said first control channel, and further comprising wherein said first system is configured to;

establish said first control channel between said first system, and said second system, in advance of processing said one or more first incoming audio frequency signals, comprising wherein said first system is configured to;

set up at least one voice trunk as soon as said first system and said second system are switched on, wherein said first control channel is ready when a first call is made, providing for faster call connect, and greater efficiency than absent said establishing.

17. The system according to claim 1, comprising:

wherein said first system configured to create comprises using said first control channel, and further comprising wherein said first system is configured to;

wait for receipt of a first call, and

upon said receipt of said first call,

establish said first control channel between said first system, and said second system, based on real time call requirements of said first call, to allow processing said one or more first incoming audio frequency signals; and

wherein said first system is configured to comprise;

wherein said first control channel comprises, depending on preprogrammed criteria, to either one of;

become permanently established;

or

become temporarily established.

18. The system according to claim 1, further comprising wherein said first system and said second system are configured to:

connect or couple said first system and said second system simultaneously to one another, creating a mesh network.

19. The system according to claim 1, wherein the packet data network comprises any one or more of:

a wired network;

a wireless network;

an optical network;

a fiber network;

a satellite network;

a 4G wireless network;

a 5G wireless network;

a 6G wireless network;

an n-G wireless network;

a Long-Term Evolution (LTE) network;

a wireless LTE network;

a GSM/EDGE network;

a UMTS/HSPA network;

a CDMA2000 network;

a 4G LTE network;

a 3G or 3GPP network;

a WiMAX network;

an evolved high speed packet access network;

an LTE Advanced network;

a WiMAX-Advanced network;

a True 4G network;

an IMT-Advanced network;

an LTE Time-Division Duplex (LTE-TDD) network;

an LTE Frequency-Division Duplex (LTE-FDD) network;

a Voice over LTE (VoLTE) network;

a Circuit-switched fallback (CSFB) network;

a Simultaneous voice and LTE (SVLTE) network;

or

a Single Radio Voice Call Continuity (SRVCC) network.

20. A method of transmitting and receiving audio frequency signals comprising PSTN-compatible, or voice over IP (VoIP) signals, or high definition voice signals over a packet data network wherein the packet data network comprises a software defined wide area network (SDWAN), the method comprising:

receiving, by at least one processor of a first system, one or more first incoming audio frequency signals,

wherein said at least one processor of said first system comprises at least one or more of;

at least one application-specific integrated circuit (ASIC),

at least one chipset,

at least one processor,

at least one microprocessor,

at least one logic circuit,

at least one quantum computer,

at least one multi-core processor,

at least one quad core processor,

at least one programmable logic device (PLD),

at least one field programmable gate arrays (FPGA),

at least one information appliance,

at least one system on a chip (SOC),

at least one multi-processor,

at least one computer processor,

at least one graphics processor, or

at least one digital signal processor (DSP),

said at least one first processor;

digitizing, by said at least one processor of said first system, said one or more first incoming audio frequency signals according to an audio standard obtaining first digitized audio frequency signals;

segmenting, by said at least one processor of said first system, said first digitized audio frequency signals to generate one or more first sequences of audio signal samples, each of said one or more first sequences of audio signal samples comprising a first preprogrammed sample size;

compressing, by said at least one processor of said first system, said each of said one or more first sequences of audio signal samples according to a first preprogrammed set of rules comprising using a first preprogrammed compression algorithm, producing a first sequence of one or more strings of processed samples, responsive to determining that said each of said one or more first sequences of audio signal samples is determined to meet a first preprogrammed criteria for compression according to said first preprogrammed set of rules;

accumulating, by said at least one processor of said first system, said first sequence of said one or more strings of processed samples creating a first group of samples ready for transmission according to a first aggregation threshold defined by a second preprogrammed set of rules;

creating a first outgoing digital message from said first group of samples ready for transmission according to a forwarding threshold defined by a third preprogrammed set of rules, using at least one of;

a first pre-defined data link protocol, or

a first control channel;

transmitting, by said at least one processor of said first system, said first outgoing digital message over the software defined wide area network to a second system;

receiving, by said at least one processor of said first system, from the software defined wide area network, and interpreting, by said at least one processor of said first system, a first incoming digital message from the second system;

processing, by said at least one processor of said first system, said first incoming digital message into one or more incoming digital audio stream samples according to said third pre-programmed set of rules comprising said first pre-defined data link protocol, or said first control channel, or a second pre-defined data link protocol, or a second control channel; and

accumulating and processing said one or more incoming digital audio stream samples according to a fourth preprogrammed set of rules comprising being configured to;

decompressing any compressed of said one or more incoming digital audio stream samples according to a first preprogrammed decompression algorithm defined by said fourth preprogrammed set of rules into decompressed digital audio samples;

accumulating said first decompressed digital samples into a first buffer according to a first jitter buffer threshold, wherein said first jitter buffer threshold is defined by said fourth preprogrammed set of rules; and

regenerating, by said at least one processor of said first system, first outgoing audio frequency signals based on said first decompressed digital audio samples.

21. The method of claim 20, further comprising at least one or more of:

wherein the software defined wide area network comprises an Internet protocol (IP) based network;

or

wherein the software defined wide area network comprises a LTE software defined wide area network;

or

providing, by said at least one processor of said first system, an interface to a voice over Internet protocol (VoIP) packet system or High Definition Voice system.

22. The method of claim 20, further comprising:

receiving, by at least one processor of the second system, one or more second incoming audio frequency signals,

wherein said at least one processor of the second system comprises at least one or more of;

at least one application-specific integrated circuit (ASIC),

at least one chipset,

at least one processor,

at least one microprocessor,

at least one logic circuit,

at least one quantum computer,

at least one multi-core processor,

at least one quad core processor,

at least one programmable logic device (PLD),

at least one field programmable gate arrays (FPGA),

at least one information appliance,

at least one system on a chip (SOC),

at least one multi-processor,

at least one computer processor,

at least one graphics processor, or

at least one digital signal processor;

digitizing, by said at least one processor of the second system, said one or more second incoming audio frequency signals according to an audio standard obtaining second digitized audio frequency signals;

segmenting, by said at least one processor of the second system, said second digitized audio frequency signals generating one or more second sequences of audio signal samples, each of said one or more second sequences of said audio signal samples comprising said first preprogrammed sample size or a second preprogrammed sample size;

compressing said each of said one or more second sequences of audio signal samples according to said first preprogrammed set of rules or a fifth preprogrammed set of rules comprising said first preprogrammed compression algorithm or a second preprogrammed compression algorithm, producing a second sequence of one or more strings of processed samples, responsive to determining that said each of said one or more second sequences of audio signal samples is determined to meet said first or a second preprogrammed criteria for compression according to said first or said fifth preprogrammed set of rules;

accumulating, by said at least one processor of the second system, said second sequence of said one or more strings of processed samples, creating a second group of samples ready for transmission according to said first aggregation threshold, or a second aggregation threshold defined by a sixth preprogrammed set of rules;

creating a second outgoing digital message from said second group of samples ready for transmission according to said forwarding threshold, or a second forwarding threshold defined by a seventh preprogrammed set of rules, wherein said second outgoing digital message for transmission is created using said first or said second pre-defined data link protocol, or said first or said second control channel, or a third pre-defined data link protocol, or a third control channel;

transmitting, by said at least one processor of the second system, said second outgoing digital message over the software defined wide area network to said first system;

receiving, by said at least one processor of the second system, at the second system from the software defined wide area network, and interpreting, by said at least one processor of the second system, a second incoming digital message from said first system; and

processing, by said at least one processor of the second system, said second incoming digital message into one or more second incoming digital audio stream samples according to said third preprogrammed set of rules or an eighth preprogrammed set of rules comprising said first, said second, or said third pre-defined data link protocol, or said first, said second, or said third control channel, or a fourth pre-defined data link protocol or a fourth control channel;

accumulating, by said at least one processor of the second system, and processing said one or more second incoming digital audio stream samples according to said fourth preprogrammed set of rules, or a ninth preprogrammed set of rules;

decompressing, by said at least one processor of the second system, said one or more of said second incoming digital audio sample streams according to a second preprogrammed decompression algorithm as defined by said fourth or said ninth preprogrammed set of rules into second decompressed digital audio samples;

accumulating, by said at least one processor of the second system, said second decompressed digital audio samples into a second buffer according to a second jitter buffer threshold, wherein said second jitter buffer threshold is defined by said fourth or said ninth preprogrammed set of rules; and

regenerating, by said at least one processor of the second system, second outgoing audio frequency signals based on said second decompressed digital audio samples.

23. The method of claim 22, wherein said first system and the second system are at least one of:

at a single location, or

at different locations.

24. The method of claim 22, further comprising:

using at least one of;

predetermined information, or

learned information, or

preconfigured information,

in determining at least one of said first, or said second, or said third, or said fourth preprogrammed rules to apply to processing, by said at least one processor of said first system or the second system, said first or said second signal information and forwarding of messages between said first and the second systems.

25. The method of claim 22, further comprising:

using a control channel comprising at least one of;

using an in-band control channel, or

using an out-of-band control channel,

said using said in-band, or out-of-band control channel comprising;

remotely managing, by said at least one processor of said first system or the second system, said first and the second systems, and

wherein said using said control channel comprises;

providing, by said at least one processor of said first system or the second system, communications performing at least one of;

providing, by said at least one processor of said first system or the second system, a monitoring function;

or

providing, by said at least one processor of said first system or the second system, a control function;

or

determining, by said at least one processor of said first system or the second system, real time diagnostic information;

or

determining, by said at least one processor of said first system or the second system, status information;

or

determining, by said at least one processor of said first system or the second system, ancillary information.

26. The method of claim 20, further comprising:

compressing, and

decompressing,

by said at least one processor of at least one of said first system or the second system, at least one of;

PSTN-compatible, or

VoIP-compatible, or

High Definition Voice-compatible

audio signals.

27. The method according to claim 20, wherein said creating comprises using said first control channel, and further comprising:

establishing said first control channel between said first system, and said second system, in advance of processing said one or more first incoming audio frequency signals, comprising;

setting up at least one voice trunk as soon as said first system and said second system are switched on, wherein said first control channel is ready when a first call is made, providing for faster call connect, and greater efficiency than absent said establishing.

28. The method according to claim 20, wherein said creating comprises using said first control channel, and further comprising:

waiting for receipt of a first call, and

upon said receipt of said first call,

establishing said first control channel between said first system, and said second system, based on real time call requirements of said first call, to allow processing said one or more first incoming audio frequency signals; and

wherein said first control channel comprises, depending on

preprogrammed criteria, either one of;

becoming permanently established;

or

becoming temporarily established.

29. The method according to claim 20, further comprising:

connecting or coupling said first system and said second system, simultaneously to one another, creating a mesh network.

30. The method according to claim 20, wherein the packet data network comprises any one or more of:

a wired network;

a wireless network;

an optical network;

a fiber network;

a satellite network;

a 4G wireless network;

a 5G wireless network;

a 6G wireless network;

an n-G wireless network;

a Long-Term Evolution (LTE) network;

a wireless LTE network;

a GSM/EDGE network;

a UMTS/HSPA network;

a CDMA2000 network;

a 4G LTE network;

a 3G or 3GPP network;

a WiMAX network;

an evolved high speed packet access network;

an LTE Advanced network;

a WiMAX-Advanced network;

a True 4G network;

an IMT-Advanced network;

an LTE Time-Division Duplex (LTE-TDD) network;

an LTE Frequency-Division Duplex (LTE-FDD) network;

a Voice over LTE (VoLTE) network;

a Circuit-switched fallback (CSFB) network;

a Simultaneous voice and LTE (SVLTE) network;

or

a Single Radio Voice Call Continuity (SRVCC) network.

31. A nontransitory computer machine-readable medium that provides instructions, which when executed by a computer processor of a computing platform, causes the computing platform to perform operations comprising a method offor transmitting and receiving audio frequency signals comprising PSTN-compatible, or voice over IP (VoIP) signals, or high definition voice signals over a packet data network wherein the packet data network comprises a software defined wide area network (SDWAN), the method comprising:

receiving, by at least one processor of a first system, one or more first incoming audio frequency signals,

wherein said at least one processor of said first system comprises at least one or more of;

at least one application-specific integrated circuit (ASIC),

at least one chipset,

at least one processor,

at least one microprocessor,

at least one logic circuit,

at least one quantum computer,

at least one multi-core processor,

at least one quad core processor,

at least one programmable logic device (PLD),

at least one field programmable gate arrays (FPGA),

at least one information appliance,

at least one system on a chip (SOC),

at least one multi-processor,

at least one computer processor,

at least one graphics processor, or

at least one digital signal processor (DSP);

digitizing, by said at least one processor of said first system, said one or more first incoming audio frequency signals according to an audio standard obtaining first digitized audio frequency signals;

segmenting, by said at least one processor of said first system, said first digitized audio frequency signals to generate one or more first sequences of audio signal samples, each of said one or more first sequences of audio signal samples comprising a first preprogrammed sample size;

compressing, by said at least one processor of said first system, said each of said one or more first sequences of audio signal samples according to a first preprogrammed set of rules comprising using a first preprogrammed compression algorithm, producing a first sequence of one or more strings of processed samples, responsive to determining that said each of said one or more first sequences of audio signal samples is determined to meet a first preprogrammed criteria for compression according to said first preprogrammed set of rules;

accumulating, by said at least one processor of said first system, said first sequence of said one or more strings of processed samples creating a first group of samples ready for transmission according to a first aggregation threshold defined by a second preprogrammed set of rules;

creating a first outgoing digital message from said first group of samples ready for transmission according to a forwarding threshold defined by a third preprogrammed set of rules, using at least one of;

a first pre-defined data link protocol, or

a first control channel;

transmitting, by said at least one processor of said first system, said first outgoing digital message over the software defined wide area network to a second system;

receiving, by said at least one processor of said first system, from the software defined wide area network, and interpreting, by said at least one processor of said first system, a first incoming digital message from the second system;

processing said first incoming digital message into one or more incoming digital audio stream samples according to said third pre-programmed set of rules comprising said first pre-defined data link protocol, or said first control channel, or a second pre-defined data link protocol, or a second control channel; and

accumulating and processing said one or more incoming digital audio stream samples according to a fourth preprogrammed set of rules comprising being configured to;

decompressing any compressed of said one or more incoming digital audio stream samples according to a first preprogrammed decompression algorithm defined by said fourth preprogrammed set of rules into decompressed digital audio samples;

accumulating said first decompressed digital samples into a first buffer according to a first jitter buffer threshold, wherein said first jitter buffer threshold is defined by said fourth preprogrammed set of rules; and

regenerating, by said at least one processor of said first system, first outgoing audio frequency signals based on said first decompressed digital audio samples.

32. The nontransitory computer machine-readable medium of claim 31, wherein the method comprises:

performing functions of said first system and the second system at least one of;

at a single location, or at different locations.

33. The nontransitory computer machine-readable medium of claim 31, wherein the software defined wide area network comprises an LTE software defined wide area network.

34. The nontransitory computer machine-readable medium of claim 31, wherein the method further comprises:

receiving, by at least one processor of the second system, one or more second incoming audio frequency signals;

digitizing, by said at least one processor of the second system, said one or more second incoming audio frequency signals according to an audio standard obtaining second digitized audio frequency signals;

segmenting, by said at least one processor of the second system, said second digitized audio frequency signals generating one or more second sequences of audio signal samples, each of said one or more second sequences of said audio signal samples comprising said first preprogrammed sample size or a second preprogrammed sample size;

compressing said each of said one or more second sequences of audio signal samples according to said first preprogrammed set of rules or a fifth preprogrammed set of rules comprising said first preprogrammed compression algorithm or a second preprogrammed compression algorithm, producing a second sequence of one or more strings of processed samples, responsive to determining that said each of said one or more second sequences of audio signal samples is determined to meet said first or a second preprogrammed criteria for compression according to said first or said fifth preprogrammed set of rules;

accumulating, by said at least one processor of the second system, said second sequence of said one or more strings of processed samples, creating a second group of samples ready for transmission according to said first aggregation threshold, or a second aggregation threshold defined by a sixth preprogrammed set of rules;

creating a second outgoing digital message from said second group of samples ready for transmission according to said forwarding threshold, or a second forwarding threshold defined by a seventh preprogrammed set of rules, wherein said second outgoing digital message for transmission is created using said first or said second pre-defined data link protocol, or said first or said second control channel, or a third pre-defined data link protocol, or a third control channel;

transmitting, by said at least one processor of the second system, said second outgoing digital message over the software defined wide area network to said first system;

receiving, by said at least one processor of the second system, at the second system from the software defined wide area network, and interpreting, by said at least one processor of the second system, a second incoming digital message from said first system; and

processing, by said at least one processor of the second system, said second incoming digital message into one or more second incoming digital audio stream samples according to said third preprogrammed set of rules or an eighth preprogrammed set of rules comprising said first, said second, or said third pre-defined data link protocol, or said first, said second, or said third control channel, or a fourth pre-defined data link protocol or a fourth control channel;

accumulating, by said at least one processor of the second system, and processing said one or more second incoming digital audio stream samples according to said fourth preprogrammed set of rules, or a ninth preprogrammed set of rules;

decompressing, by said at least one processor of the second system, said one or more of said second incoming digital audio sample streams according to a second preprogrammed decompression algorithm as defined by said fourth or said ninth preprogrammed set of rules into second decompressed digital audio samples;

accumulating, by said at least one processor of the second system, said second decompressed digital audio samples into a second buffer according to a second jitter buffer threshold, wherein said second jitter buffer threshold is defined by said fourth or said ninth preprogrammed set of rules; and

regenerating, by said at least one processor of the second system, second outgoing audio frequency signals based on said second decompressed digital audio samples.

35. The nontransitory computer addressable medium according to claim 31, wherein the method comprises:

wherein said creating comprises using said first control channel, and further comprising;

establishing said first control channel between said first system, and said second system, in advance of processing said one or more first incoming audio frequency signals, comprising;

setting up at least one voice trunk as soon as said first system and said second system are switched on, wherein said first control channel is ready when a first call is made, providing for faster call connect, and greater efficiency than absent said establishing.

36. The nontransitory computer addressable medium according to claim 31, wherein the method comprises:

wherein said creating comprises using said first control channel, and further comprising;

waiting for receipt of a first call, and

upon said receipt of said first call,

establishing said first control channel between said first system, and said second system, based on real time call requirements of said first call, to allow processing said one or more first incoming audio frequency signals and

wherein said first control channel comprises, depending on

preprogrammed criteria, either one of;

becoming permanently established;

or

becoming temporarily established.

37. The nontransitory computer accessible medium according to claim 31, wherein the method further comprises:

connecting or coupling said first system and said second system, simultaneously to one another, creating a mesh network.

38. The nontransitory computer accessible medium according to claim 31, wherein the packet data network comprises any one or more of:

a wired network;

a wireless network;

an optical network;

a fiber network;

a satellite network;

a 4G wireless network;

a 5G wireless network;

a 6G wireless network;

an n-G wireless network;

a Long-Term Evolution (LTE) network;

a wireless LTE network;

a GSM/EDGE network;

a UMTS/HSPA network;

a CDMA2000 network;

a 4G LTE network;

a 3G or 3GPP network;

a WiMAX network;

an evolved high speed packet access network;

an LTE Advanced network;

a WiMAX-Advanced network;

a True 4G network;

an IMT-Advanced network;

an LTE Time-Division Duplex (LTE-TDD) network;

an LTE Frequency-Division Duplex (LTE-FDD) network;

a Voice over LTE (VoLTE) network;

a Circuit-switched fallback (CSFB) network;

a Simultaneous voice and LTE (SVLTE) network;

or

a Single Radio Voice Call Continuity (SRVCC) network.