On 14.02.22 19:23, Juha Heinanen wrote:
Daniel-Constantin Mierla writes:
WebSocket (for WebRTC) * send SIP requests of any type (e.g., INFO, SUBSCRIBE, NOTIFY, …)
One usage example that could ease the testing of Kamailio is initiating registrations or simulating calls over WebSocket without the need of having a JavaScript soft phone application running in a web browser.
Thanks for the tool. Regarding SIP over WebSocket, baresip supports WebSocket transport in all platforms.
baresip is more like a proper SIP phone (which is great and I use it for such purpose), but I don't think it has the option to "forge" any kind of SIP request. The sipexer is a result of not having enough time to (fully understand and then) code C/C++ for sipsak to add websocket support (plus a few other like IPv6, more TLS flexibility).
I wrote a couple of years ago wsctl to be able to do testing over websocket from cli, I don't think baresip had support for websocket at that time, anyhow my need was mainly to be able to reproduce by sending SIP traffic from a previous capture) and a few months ago I decided to start a more sip-oriented tool written in golang, considering is faster development due to embedded tls support and easier websocket integration (also hoping that contributions will be easier in golang than c/c++ nowadays from the new generation).
sipexer has to be seen as a sip cli tool, not as a sip softphone, there is no media/audio support.
Cheers, Daniel