14 Feb
2022
14 Feb
'22
9:57 p.m.
Certainly, but 90% of the various use-cases are covered by the invite scenario. Extensive
compatibility with various CI tooling isn’t really required in my mind; as long as it can
return positive or negative values depending on the outcome of the SIP request, it’s
perfect.
The real value is in the fact that it’s a true CLI tool, and the ability to formulate
misshapen requests using Go templates. That’s beautiful!
Another great thing is that you appear to have exposed your ad hoc SIP parser as a module,
which means it could potentially be imported and used in other tools.
—
Sent from mobile, with due apologies for brevity and errors.
On Feb 14, 2022, at 1:51 PM, Daniel-Constantin Mierla
<miconda(a)gmail.com> wrote:
Probably it requires some hammering to make it compatible with various
CI pipelines, I tried to make a mode for nagious plugin, but coding in
golang should make it easy to adapt/enhance.
I plan to add a few more common scenarios for session testing. Right now
can do register-wait-unregister and invite/200ok-ack-wait-bye.
One that is my to-do is to register two users and make a call between
them. Another one would be to register and wait for calls, so another
sipexer instance can be used for register and initiate calls.
Writing the sip traffic in a pcap file is something that I would like to
add as well.
Cheers,
Daniel
On 14.02.22 19:27, Alex Balashov wrote:
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
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
_______________________________________________
Kamailio (SER) - Development Mailing List
sr-dev(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev
--
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/
On Feb
14, 2022, at 1:23 PM, Juha Heinanen <jh(a)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
_______________________________________________
Kamailio (SER) - Development Mailing List
sr-dev(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev