Glorious!!!!!
Thanks a lot Daniel, very very very much appreciated!!!
-giovanni
On Mon, Feb 14, 2022 at 7:42 PM Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
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)
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https://www.asipto.com/sw/kamailio-advanced-training-online/
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Sincerely,
Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18