[SR-Users] [sr-dev] Info: sipexer v1.0.0 - sip cli tool

[Daniel-Constantin Mierla]

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
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