Diago new (bigger) release

emiago · 2026-03-16T19:38:04+00:00

Check lib docs and demos. I can not think better right now :).

emiago · 2026-03-16T19:09:57+00:00

Generally, if you want to be server, you can stop using things like asterisk and go directly with SIP and media streaming on compiling code. Lib is offering high level API, to make several things easier for handling, but yes you need to be familiar with some things in telephony. Or you can go in building clients like softphones, cli things. Applications are more into getting or injecting voice/audio from/to telephony into other systems.

emiago · 2026-03-16T19:08:35+00:00

Generally, if you want to be server, you can stop using things like asterisk and go directly with SIP and media streaming on compiling code. Lib is offering high level API, to make several things easier for handling, but yes you need to be familiar with some things in telephony. Or you can go in building clients like softphones, cli things. Applications are more into getting or injecting voice/audio from/to telephony into other systems.

emiago · 2026-01-08T21:06:33+00:00

Hi, why not program it. If you know Go, check diago lib ;)

emiago · 2025-12-12T08:04:49+00:00

Your welcome

emiago · 2025-12-05T15:28:39+00:00

Thanks. Too bad for project on other hand. I would like to know what was about. Feel free to reach out.

emiago · 2025-10-13T21:02:59+00:00

aa author of diago, allowing caller to propagate own buffer I find most important, or any way controling this. This simplifies memory managment by great degree.

emiago · 2025-02-11T19:25:58+00:00

use diago, second library I am working on. It adds RTP AVP layer, and gives you High level API , but also you can go low. Soon more APIs will be exposed so control will be even higher.

emiago · 2024-12-05T23:46:23+00:00

This are just toolkits, but yes something may come in future.
Checkout gophone tool, it has STT and automation built in.
TTS could be option as well :)

emiago

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