Help with Audio Conversions #3
Description
Hello @mscdex,
I'm trying to follow your example (sendg711.js) on streaming RTP-based audio to a Cisco phone. I used https://g711.org/ to convert one of my .wav using the "Standard Definition 16-bit WAV (8Khz, Mono, 16-Bit PCM)" option. This produced the clearest audio on the phone, whereas the rest of the conversions were intercombined with static or completely unintelligible.
However, I'm having some issues with receiving the audio on my phone. Using the default code:
intvl = setInterval(writeData, 20);
yields the audio streaming for about 2 seconds, no audio for 2 seconds, and the audio continuing intermittently so on. If I decrease the param of 20 to 8 (for example), there are no more gaps of silence but the audio appears to 'overlap' during the streaming process. Increasing past 20 also respectively increases the gaps.
Do you have any tips on what the problem might be or how I should better convert my audio file?