CouthIT’s Voice over IP library is a bundled collection of software components that allows to make a end-to-end voice calls over IP networks. This module provides an integrated framework for voice communications over Internet clouds while ensuring maximum voice quality, minimizing distortions such as echo and background noise, intelligently  managing the network jitter experienced by IP packets for protection against network impairments such as latency, out of order arrival of packets and packet losses. The implementation is targeted for VOIP Phone,  Media Gateways, streaming media applications and voice messaging.

  • Optimized ASM/C implementation.
  • Re-entrant implementation.
  • C-callable APIs
  • Support for 8 KHz and 16 KHz sampling frequency.
  • Supports G.711, G.729AB, G.726 and G.723.1A speech codecs (optional support for other speech codecs).
  • Support for VAD/DTX/CNG with G.711 and G.726 to reduce channel payload during silence periods.
  • Support for arbitrary packet size (default set to 20 ms).
  • Noise suppression enhances the speech signal by suppressing stationary and non-stationary background noises.
  • AGC adaptively maintains the dynamic range of a speech signal without amplifying the non speech portions.
  • Acoustic echo canceler removes unwanted echo from the speech signals while providing near full duplex communications.
  • Provides packet loss concealment in case of missing packets to improve the speech quality.
  • Jitter buffer compensates for IP based network impairments such as latency, out of order arrival of packets.
  • Support for static or adaptive jitter buffer; maximum jitter delay configurable at initialization time.
  • Optimized for low memory foot print and low complexity.
  • Tested using a large database of test vectors simulating real use scenarios.
  • Tested over Internet cloud for real time performance.
  • Evaluated Noise suppression modules using babble, car, street, and a combination of stationary and non-stationary background noises.
  • Evaluated AGC Module for different speech levels from -15dBov to -40dBov
  • Tested using network simulator implementation for out of order arrival of packets.
  • Tested using random arrival time distribution models including linear, uniform, Gaussian and Poisson distribution.
  • Performance benchmarked using objective evaluation measures such as log likelihood ratio, segmental SNR and weighted spectral slope.
  • Tested by integrating with open-source RTP/RTCP and SIP and registering with Asterisk PBX server.
  • Tested for interoperability using multiple open-source and proprietary VoIP client software.
  • Tested for graceful exit in case of errors or exception.
  • Tested for Input buffer corruption
  • Tested for I/O buffer alignment requirements
  • Tested for multi-instance implementation.
  • Module is fully interruptible.
  • Tested for 100% code coverage
  • Range validation of all API parameters.
  • Code validated on ARM926EJ-S (OMAP L138) platform running Open Embedded Linux.

ARM9E, ARM11, Cortex-A8, and Cortex-A9

For datasheet with resource usage details