• About Us

    About Us

    Consilient develops optimised software algorithms involved in voice, data, fax and video communications. We also provide integrated solutions for the embedded and networked markets.

  • Products
    Multimedia Codecs
    VoLTE Voice Engine
    Voice Quality Enhancement
    Acoustic Beamforming
    VoIP Call Decoder
    Embedded SIP based VoIP
    SIP SoftModem Server (SSMS)

    Multimedia Codecs

    In today's rapidly advancing technological world, high-quality communication and multimedia experiences are essential. Consilient offers a superior range of voice, audio, and video codecs designed to meet the highest performance, reliability, and efficiency standards.

    Know More

    VoLTE Voice Engine

    In the fast-evolving world of mobile communication, delivering clear, high-quality voice calls over LTE networks has become a fundamental necessity. Consilient's VoLTE Voice Engine is at the forefront of this technological revolution, offering a superior voice-over LTE (VoLTE) service solution.

    Know More

    Voice Quality Enhancement

    In today's digital age, clear and effective communication is paramount. Whether for business meetings, virtual conferences, or personal calls, superior voice quality can significantly impact the user experience. Consilient's Voice Quality Enhancement solutions are designed to address common audio challenges.

    Know More

    Acoustic Beamforming

    At Consilient, we are at the forefront of audio innovation, bringing you state-of-the-art acoustic microphone beamforming software that transforms how you capture and experience sound. Our cutting-edge acoustic beamforming products are designed to provide unparalleled clarity and precision.

    Know More

    VoIP Call Decoder

    Consilient's VoIP Call Decoder is an innovative solution designed to meet the growing demands of Lawful Interception and communication management. This state-of-the-art product is engineered to handle voice, fax, and video data, providing unparalleled capabilities for VoIP decoding.

    Know More

    Embedded SIP based VoIP

    In the ever-evolving landscape of embedded systems, integrating Voice over Internet Protocol (VoIP) has opened new avenues for communication technology. Consilient has developed a cutting-edge Embedded SIP based VoIP (VoIP) product that offers a compact and powerful solution tailored for microcontroller (MCU).

    Know More

    SIP SoftModem Server (SSMS)

    The demand for efficient and scalable communication solutions has never been higher in today's digital age. Consilient's SIP SoftModem Server (SSMS) is designed to meet this demand, offering a robust and versatile Cloud SIP-based SoftModem server solution.

    Know More
  • Services
  • Resources
  • Contact Us

AMR Codec

Overview

NB-AMR Codec - Narrowband Adaptive Multi-Rate Speech Codec for Mobile & VoIP

The NB-AMR codec, short for Narrowband Adaptive Multi-Rate codec, is a speech codec standardized by 3GPP for encoding narrowband speech at various bitrates. It is widely used in GSM and UMTS networks to provide efficient and intelligible voice communication over bandwidth-limited channels. The codec supports a frequency range up to 3.4 kHz, suitable for traditional telephony voice quality.

Technical Specifications - NB-AMR Codec

Parameter Specification
Full Name Narrowband Adaptive Multi-Rate (NB-AMR) Codec
Standardized By 3GPP (ETSI TS 26.071, 26.101)
Bandwidth Narrowband (50 Hz to 3400 Hz)
Sampling Rate 8 kHz
Bit Rates Supported 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2, 12.2 kbps
Frame Size 20 ms per frame
Codec Delay Approx. 25 - 35 ms
Frame Structure 16-bit encoded frames with CRC error detection
Algorithm Algebraic Code Excited Linear Prediction (ACELP)
Voice Activity Detection Yes (Includes Discontinuous Transmission support)
Error Resilience Robust against transmission errors and packet loss
Complexity Moderate - suitable for mobile platforms

Applications Areas of NB-AMR Codec

Mobile & Cellular Communication

  • Standard voice codec for GSM and UMTS networks
  • Widely used in 2G/3G circuit-switched networks

IP and VoIP Applications

  • SIP Clients and VoIP systems
  • WebRTC, conferencing software
  • Softphone and collaboration tools

Multimedia & Messaging

  • MMS support (.amr file format)
  • Voice recordings in 3GP, AMR file containers

Consumer & Playback Devices

  • Smartphones, feature phones
  • Embedded voice applications
  • Media playback (support via Android, FFmpeg, etc.)

Performance Summary - LC3/LC3Plus Codec

Bitrate (kbps) Speech Quality (MOS-LQO) Use Case
4.75 ~3.2 Low-bitrate, poor network conditions
7.95 ~3.8 Moderate quality, efficient transmission
12.2 ~4.2 Maximum quality (standard telephony voice)

Supported Platforms

  • Operating Systems: Linux, Android, iOS, Embedded RTOS
  • Architectures: ARM Cortex-A/R/M, NEON/SIMD, x86/x64, MIPS DSP, Andes DSP, Verisilicon ZSP

Choose the NB-AMR Codec?

High speech quality over mobile and VoIP networks

Bandwidth efficiency and scalable bitrate control

Resilience to network conditions and errors

Mature, proven adoption in GSM/UMTS systems globally

Support for real-time voice and stored multimedia use cases

Low-complexity suitable for constrained hardware

Excellent efficiency and resilience for circuit-switched networks

References & Standards - NB-AMR Codec

  • ETSI TS 126.071 -AMR Speech Codec: General Description
  • ETSI TS 126.090 - AMR Codec Transcoding functions
  • ETSI TS 126.091 - AMR Codec Error concealment of lost frames
  • ETSI TS 126.092 -AMR Codec Comfort noise aspects
  • ETSI TS 126.093 -AMR Codec Source controlled rate operation
  • ETSI TS 126.094 -AMR Codec Voice activity detector
  • IETF RFC 4867 -RTP Payload Format for AMR and AMR-WB Codecs
Overview

WB-AMR Codec (Wideband Adaptive Multi-Rate) - Data Sheet

The WB-AMR codec, short for Wideband Adaptive Multi-Rate codec, is a speech audio codec designed to encode wideband speech signals for voice communication over various transmission systems. The WB-AMR codec supports higher audio quality than narrowband codecs by extending the frequency range up to 7 kHz, enhancing intelligibility and naturalness.

Technical Specifications - WB-AMR Codec

Parameter Specification
Full Name Wideband Adaptive Multi-Rate (WB-AMR) Codec
Standardized By 3GPP (ETSI TS 26.171, 26.190)
Bandwidth Wideband (50 Hz to 7000 Hz)
Sampling Rate 16 kHz
Bit Rates Supported 6.60, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05, 23.85 kbps
Frame Size 20 ms per frame
Codec Delay Approx. 34-45 ms
Frame Structure 16-bit encoded frames with error detection
Algorithm Algebraic Code Excited Linear Prediction (ACELP)
Voice Activity Detection Yes (with Discontinuous Transmission support)
Error Resilience Robust against transmission errors and packet loss
Complexity Moderate - suitable for mobile platforms

Applications Areas of WB-AMR Codec

Mobile & Cellular Communication

  • HD Voice for GSM, UMTS, LTE, WCDMA
  • VoLTE and VoWiFi support using WB AMR codec
  • Voice over 5G applications

IP and VoIP Applications

  • SIP and VoIP clients
  • WebRTC, conferencing software
  • Softphone and collaboration tools

Multimedia & Messaging

  • Enhanced voice recording in .awb format
  • MMS support in 3GP containers
  • Push-to-Talk, RCS messaging

Consumer & Playback Devices

  • Smartphones, feature phones
  • Embedded voice applications
  • Media playback (support via Android, FFmpeg, etc.)

Performance Summary - WB-AMR Codec

Bitrate (kbps) Speech Quality (MOS-LQO) Use Case
6.60 ~3.6 Low-bitrate streaming, low-bandwidth
12.65 ~4.0 Balanced quality/bitrate
23.85 ~4.5 Maximum speech quality (HD voice)

Supported Platforms

  • Operating Systems: Linux, Android, iOS, Embedded RTOS
  • Architectures: ARM Cortex-A/R/M, NEON/SIMD, x86/x64, MIPS DSP, Andes DSP, Verisilicon ZSP

Why choose the WB-AMR Codec?

High speech quality over mobile and VoIP networks

Bandwidth efficiency and scalable bitrate control

Resilience to network conditions and errors

Proven adoption in 3GPP, VoLTE, and HD voice systems

Support for real-time voice and stored multimedia use cases

References & Standards - WB-AMR Codec

  • ETSI TS 126.171 -AMR-WB Speech Codec: General Description
  • ETSI TS 126.190 -AMR-WB Codec Transcoding functions
  • ETSI TS 126.191 -AMR-WB Codec Error concealment of lost frames
  • ETSI TS 126.192 -AMR-WB Codec Comfort noise aspects
  • ETSI TS 126.193 -AMR-WB Codec Source controlled rate operation
  • ETSI TS 126.194 -AMR-WB Codec Voice activity detector
  • 3GPP TS 26.201 -AMR-WB Speech Codec Frame Format
  • IETF RFC 4867 - RTP Payload Format for AMR and AMR-WB Codecs

1. What is the AMR Codec, and what does Consilient Technologies offer?

The Adaptive Multi-Rate (AMR) Codec is a speech compression standard originally developed for GSM and 3G mobile networks. Consilient Technologies provides production-ready AMR encoder/decoder libraries — both Narrowband (NB-AMR) and Wideband (WB-AMR) variants — optimized for integration into telecom infrastructure, VoIP platforms, and embedded devices. Our implementation focuses on computational efficiency, standards compliance, and ease of integration into existing audio pipelines.

2. What is the difference between NB-AMR and WB-AMR, and when should I use each?

NB-AMR operates at 4.75–12.2 kbps and covers a frequency range up to 3.4 kHz — suitable for legacy GSM telephony and narrowband VoIP where bandwidth is constrained. WB-AMR (also known as AMR-WB or G.722.2) operates at 6.6–23.85 kbps with audio bandwidth up to 7 kHz, delivering noticeably richer voice quality. Choose NB-AMR when you need backward compatibility with older GSM/UMTS infrastructure or when operating on very low-bandwidth links. Choose WB-AMR when your network supports it and voice quality is a priority — for example, in VoLTE deployments or HD voice-enabled VoIP systems.

3. How does the AMR Codec adapt to changing network conditions?

AMR's defining feature is its adaptive bitrate mechanism. The codec can switch between multiple bitrate modes (eight for NB-AMR, nine for WB-AMR) on a frame-by-frame basis — every 20 ms. When the network signals congestion or increased error rates, the codec drops to a lower bitrate to reduce the data load, preserving call continuity. When conditions improve, it scales back up for better audio fidelity. This happens transparently, without interrupting the call. Consilient Technologies' implementation exposes configurable mode-set parameters so integrators can control which bitrate modes are permitted for their specific deployment.

4. What platforms and architectures does Consilient Technologies' AMR Codec support?

Our AMR Codec libraries are available as portable C source code and pre-optimized builds for ARM (Cortex-A and Cortex-M series), various DSP architectures, and x86/x64 platforms. We support integration into Linux, Android, RTOS environments (FreeRTOS, ThreadX, VxWorks), and bare-metal embedded systems. This makes the codec suitable for a wide range of targets — from cloud-based media servers to battery-powered IoT voice devices.

5. How does Consilient Technologies' AMR Codec compare to EVS or Opus for my use case?

AMR is the right choice when you need compatibility with existing 2G/3G/VoLTE mobile networks, since it remains the mandatory codec for GSM and is widely deployed in VoLTE via AMR-WB. EVS (Enhanced Voice Services) offers better audio quality at equivalent bitrates and supports fullband audio, but requires 4G/5G infrastructure and has higher computational cost. Opus is ideal for internet-based applications (WebRTC, streaming) but is not part of the 3GPP mobile standard. If your primary requirement is interoperability with mobile carrier networks, AMR is typically the most practical choice.

6. What are the typical use cases for Consilient Technologies' AMR Codec?

Common deployments include: telecom infrastructure such as media gateways and session border controllers that transcode between AMR and other formats; VoLTE and VoWiFi services where AMR-WB is the standard HD voice codec; embedded communication devices like two-way radios, in-vehicle systems, and IoT endpoints with voice capability; voice recording and storage systems that need efficient compression for large call archives; and conferencing bridges that must interwork with mobile callers.

7. How does Consilient Technologies ensure low latency in its AMR Codec implementation?

AMR inherently operates with a 20 ms frame size and a look-ahead of 5 ms (NB) or 5 ms (WB), giving an algorithmic delay of 25 ms per direction. Our implementation minimizes additional processing overhead through optimized ACELP search routines and memory-efficient buffer management. For embedded targets, we offer fixed-point implementations that avoid costly floating-point operations, keeping CPU utilization low and latency predictable — critical for real-time voice applications.

8. How can my engineering team evaluate and integrate Consilient Technologies' AMR Codec?

We provide an evaluation package that includes pre-built libraries for your target platform, sample application code, API documentation, and conformance test vectors aligned with 3GPP specifications (TS 26.073 for NB-AMR, TS 26.173 for WB-AMR). Integration typically involves linking the codec library into your audio pipeline, configuring the mode set and DTX (Discontinuous Transmission) parameters, and running validation against the provided test vectors. Our engineering team offers direct integration support for enterprise customers.

to-top