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

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https://puri.sm/posts/voice-over-ip-in-gnome-calls-part-1-sip-protocol-and-libraries?mtm_campaign=adv_tech_piece&mtm_source=organic&mtm_medium=forum&mtm_content=f-VOIP_gnome_part_1

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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 197.51.100.1 come from?

In the section on RTP, it looks like the formatting and link for ā€˜opusā€™ are broken. Maybe a markup issue.

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For fun I tried

gst-launch-1.0 -v audiotestsrc ! alsasink

I just wanted to know what the test source produced.

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

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Itā€™s very easy to discover all the available gstreamer elements with ā€œgst-inspect-1.0ā€. I just did ā€œgst-inspect-1.0 | grep volumeā€ and found that thereā€™s an element called ā€œvolumeā€. Then to know the settings that element allows, do ā€œgst-inspect-1.0 volumeā€ and you can see that thereā€™s a property called ā€œvolumeā€ 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

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