What codec is Huawei Mate60pro used to make Satellite Calls?

The AMBE-2000 codec is a digital voice compression technology that can provide high-quality voice transmission at low bit rates.

It is based on the AMBE (Advanced Multi-Band Excitation) algorithm, which models the human speech production process and encodes the speech parameters into a compact bit stream.

The AMBE-2000 codec can support bit rates from 2.0 kbps to 9.6 kbps, and can adjust the speech and/or error correction rates dynamically according to the channel conditions.

The AMBE-2000 codec can also resist noise and interference effectively, making it suitable for satellite and mobile radio communications.

Some technical data about the AMBE-2000 codec are:

• It uses a 20 ms frame size and a 10 ms look-ahead buffer, resulting in a total algorithmic delay of 30 ms.
• It has a built-in 16-bit ADC and 12-bit DAC, with a sampling rate of 8 kHz and a frequency response of 300 Hz to 3400 Hz.
• It has a serial interface that supports synchronous or asynchronous modes, with a maximum baud rate of 115.2 kbps.
• It has a power supply voltage of 3.3 V and a power consumption of less than 100 mW.
• It has a package size of 28-pin SOIC (10.16 mm x 7.62 mm x 2.29 mm).

You can find more information about the AMBE-2000 codec from the following sources:

• DVSI Hardware Documentation
• DVSI AMBE-2020™ Vocoder Chips
• AMBE2000 Datasheet
• Vocoder Redux: the AMBE-2020

Is there similar VoIP codec comparing to AMBE-2000 codec ?

There are some VoIP codecs that have similar bit rates to the AMBE-2000 codec, but they may not have the same voice quality or noise resistance.

Some examples are:

• G.729: This is a standard codec for narrowband speech compression, which operates at 8 kbps and uses the Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP) algorithm. It is widely used in VoIP applications and has good performance in low-delay and low-loss scenarios.
• iLBC: This is an open-source codec for narrowband speech compression, which operates at 13.33 kbps or 15.2 kbps and uses the Block-Independent Linear Predictive Coding (BILPC) algorithm. It is designed to be robust against packet loss and has low complexity and delay.
• Opus: This is a versatile codec for both narrowband and wideband speech compression, which can operate at any bit rate from 6 kbps to 510 kbps and uses the SILK and CELT algorithms. It is optimized for interactive audio applications and can adapt to different network conditions and audio content.

You can find more information about these codecs from the following sources:

• [G.729 – Wikipedia]
• [iLBC – Wikipedia]
• [Opus (audio format) – Wikipedia]

One possible reason why satellite calls do not use VoIP codecs is that VoIP codecs require a minimum bandwidth for the call in both directions (uplink and downlink), which some satellite services may not be able to provide.

For example, depending on the codec used, VoIP may require as high as 100 kbps for the call, while some satellite services may have poor bandwidth in one direction, which can affect the quality of the call.

Another possible reason is that VoIP codecs are sensitive to latency and packet loss, which are common issues in satellite communications.

For example, the round-trip time for a satellite call can be as high as 500 ms, which can cause noticeable delays and echo in the conversation.

Moreover, the satellite signal may be affected by weather conditions, interference, or congestion, which can result in packet loss and jitter. These factors can degrade the voice quality and intelligibility of VoIP calls.

Therefore, satellite calls may use different codecs that are more suitable for the satellite environment, such as AMBE-2000, which can provide high-quality voice transmission at low bit rates and resist noise and interference effectively.