Send audio to Nodejs in a bufferred and efficient way - perhaps with WebRTC

I would like to accomplish something like this in the browser but plugin-free.

It's described in a fully functional Linux command line and in pseudocode:

$ arecord -d 5 -f dat -c 1 -f S16_LE -r 8000 |\
  speexenc - - |\
  curl -X PUT "https://api.example.com/record-sound" -T -

Pesudo-code:

1. Open the mic,
2. Open the connection to the API endpoint
3. Read bytes from the mic on the rate of 8000 Hz (wave pcm signed 16-bit little endian mono 8000 Hz)
4. Encode the bytes to Speex in chunks (http://speex.org)
5. Write the byte chucks you get from Speex encoder to the webservice request body in chunks - this a lot of bandwidth is saved and bytes are sent as soon as they're available
6. When mic stops giving results (either by a timer -d 5 or stopped by the user, flush the request body and send the request.
7. Wait for JSON (or anything else) results in response body

Obviously this is a Linux command line but the algorithm have been successfully applied on:

1. Linux - using the command above
2. iOS - iPhone and iPad
3. Java2SE on Windows

I would like to achieve the same goal "efficient and fast streaming from microphone to a server" on the browser (old browsers are not my concern now). It doesn't have to be a REST server, WebSockets or whatever is possible are as good. Also Node.js based solution is prefered.

What I have tried so far:

• Successfully used a VideoIO.swf which is a Flash based solution and RTMPLite on the server to receive video. It's actually a very good solution in terms of functionality except that:
  • It requires user approval through very bad UI
  • It's not a standard way of doing web and don't allow good error detection and such
• Successfully used getUserMedia to upload the sound through recorder.js and XHR2 it works find except that:
  • It is actually slow because XHR2 needs full data before request can be made using xhr.send(data)
  • It is even slower since I could only upload wav format which is way bigger in size compared with ogg and opus
• Researched on the use of WebRTC but couldn't find any resource that mentioned anything other than peer-to-peer connections