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@gmail.com:
Cool cool cool.
On Tue, Feb 15, 2022 at 3:01 AM Alex Balashov abalashov@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@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@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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