Voice over IP in GNOME Calls Part 1: SIP Protocol and Libraries



Thanks for putting that together. Lots of useful info.

There were a few things in the section on SIP that were a bit opaque to me e.g.

  • Alice calls bob@baumeister.com but somehow it shows the call as To: bob@builders.com - I can believe that various pieces of software might map a destination, but I have my doubts that translating from German to English would be part of that. :wink:
  • In the SDP message, where does the IP address of come from?

In the section on RTP, it looks like the formatting and link for ‘opus’ are broken. Maybe a markup issue.

For fun I tried

gst-launch-1.0 -v audiotestsrc ! alsasink

I just wanted to know what the test source produced.

It’s a bit loud. So now I am wondering whether there is a pipeline element that can reduce the volume. :wink:

It’s very easy to discover all the available gstreamer elements with “gst-inspect-1.0”. I just did “gst-inspect-1.0 | grep volume” and found that there’s an element called “volume”. Then to know the settings that element allows, do “gst-inspect-1.0 volume” and you can see that there’s a property called “volume” which is a float where 0.0 is 0% and 1.0 is 100%.

So with that in mind, the following pipeline works to set the volume to 20%

gst-launch-1.0 -v audiotestsrc ! volume volume=0.2 ! alsasink