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

Wed Feb 16 10:05:22 CET 2022

Daniel,
Thanks, now we have something between sipsak and sipp.
But just small clarification on CLI parameters format. I've noticed, that
long and short version of parameter starting with single -(hyphen). Is it
something go-specific, or it could be used in more-gnu way like single
hyphen for short and double-hyphen for long?
As an example from help
  -co
        color output
  -color-message
        color SIP message output
  -color-output
        color output
  -com
        color SIP message output

And to add, support of ENV vars is a great idea! Now using separate docker
images with CI/CD looks even better!
Many thanks for your work!

As an another testing suite example: https://github.com/igorolhovskiy/volts
It's based on voip_patrol, but a bit reworked.  (Was about to make a
presentation of it on next KWC, please count me in :) )
I'm thinking of expanding it with other SIP testing tools and sipexer looks
nice from this point of view.

PS: For hyphen/double-hyphen I can try to prepare a PR, but not sure if
it's not compatibility break with already possible existing pipelines
somewhere.

вт, 15 февр. 2022 г. в 05:05, Seven Du <dujinfang at gmail.com>:

> Cool cool cool.
>
> On Tue, Feb 15, 2022 at 3:01 AM Alex Balashov <abalashov at evaristesys.com>
> wrote:
>
>> 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 at 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
>> >>
>> >>>> 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
>> >>>
>> >>> _______________________________________________
>> >>> Kamailio (SER) - Development Mailing List
>> >>> sr-dev at lists.kamailio.org
>> >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev
>> >> --
>> >> 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 at 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/
>> >
>>
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>
>
> --
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
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>
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-- 
Best regards,
Igor
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