mscdex / node-rtp Goto Github PK

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?

This code is super readable, to the point, and encapsulated perfectly..

The whole implementation is just an object with getters and setters...

sender uses setters, receiver uses getters & the constructor --

so i literally just read the datagram from the socket and throw it to the constructor...

AWESOME..

But my question is, what changes would I need to make to support rfc-2250.

Which getters and setters would need to change to support 2250?

Thanks!!

every 20 ms
intvl = setInterval(writeData, 20);
but if ((bytesRead = fs.readSync(fd, buf, 0, buf.length)) > 0) {
i think it all read 0 - 160 bytes