The idea to have such software on the hand, has been in my mind for a long time.
Started developing projects such as: tSIP, HATS, I realized that, I don't have a solution that I could use as a basis for operate with VoIP.
Freeswitch was not suitable in this case, since it had to be small, portable and ready for emedding as possible.
The Baresip, as I usually used it, didn't suit this time.
So, I decided it was time to write something that would satisfy my needs.
In addition, there are some customer tasks which also need it.
This is an open-source project and available on GitHUB .
- It should be able to work on devices with 512Mb RAM (such as: OrangePI One or MTK SoCs MT7621, as possible)
- Capable with: OpenWRT and FreeBSD
- Should allow to use as a basis for developing VoIP applications or something related with it
- SIP (must be)
- WebRTC (optional)
- SIP registrar (for local users)
- SIP gateways (for: FXO/FXS equipments and cloud PBX)
- Audio and video (as possible) calls
- Calls routing, something similar to FreeSWITCH dialplan but simpler
- Web based management interface (optional)
- JSON-RPC services for automation and integration purposes
Version 0.0.1
This is an experimental version, for testing purposes only.
Actually, it's a b2bua client with extra features. <>
Implemented capabilities:
- SIP registrar and location
- SIP gateways
- Inbound/Outbound/Intrecom calls
- Dialplan with applications
- Codecs: G711, G729
- Local users