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

[Alex Balashov]

I haven’t had a chance to dig into it just yet, but this is an incredibly exciting development, and fills a very dire gap in open-source testing tools. 

SIPp was the only real game in town and, despite some very creative efforts over the years, fundamentally is not composable: it doesn’t lend itself to headless automation or embedding in CI pipelines, and isn’t terribly useful for monitoring. The remainder is a miscellany of relatively unsophisticated or quirky tools, none of which have the flexibility you are providing here. 

Very grateful that you wrote this, and excited to try it! Thank you so much for this work!

— Alex

> On Feb 14, 2022, at 1:23 PM, Juha Heinanen <jh at tutpro.com> 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.
> 
> -- Juha
> 
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-- 
Alex Balashov | Principal | Evariste Systems LLC

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