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 -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/ _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev