[asterisk-dev] question about tests for capturing SIP messages

[Henning Westerholt]

Hello,

as part of working on [1] a test was requested to cover the new functionality.

After figuring out how the basic test suite works (previous e-mail, thanks for the fast reply), I have some questions about the best approach going forward.

The basic scenario is this:


  1.  Start Asterisk with pjsip stack
  2.  Start Sipp with a re-INVITE scenario
  3.  Capturing the SIP message flow
  4.  Checking number of returned codecs in 200 OK reply on re-INVITE from Asterisk
  5.  Tear down, Result etc..

For 1-2 I've took some existing test and adapted it, was straightforward.

For task 3 I've found some existing template: tests/channels/SIP/pcap_demo, together with lib/python/sip_message.py and lib/python/pcap_listener.py.

Unfortunately, this test seems to be skipped right now due some old issues with CentOS 6. So as expected, it's not working anymore.
I have tried to do some adaptions for python3 in the libraries, but it's still fails. I started to work on this stuff yesterday and therefore I am not the best person to fix these dependencies.

Is this library planned to be updated as well?

If not, are there other suggestions how to address this PCAP based test/check?

Thanks,

Henning

[1] https://gerrit.asterisk.org/c/asterisk/+/18990

--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://gilawa.com<https://gilawa.com/>

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