Thank you, I was aware of the regexp support in sipp but have not used it in this context. The test has been added.
Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: asterisk-dev <asterisk-dev-boun...@lists.digium.com> On Behalf Of Joshua C. Colp Sent: Thursday, September 29, 2022 2:25 PM To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Subject: Re: [asterisk-dev] question about tests for capturing SIP messages On Thu, Sep 29, 2022 at 9:15 AM Henning Westerholt <h...@gilawa.com<mailto:h...@gilawa.com>> wrote: 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? You don't need to capture the SIP message flow using pcap or anything like that. There are existing PJSIP tests which cover SDP handling and use SIPp to do so: https://github.com/asterisk/testsuite/tree/master/tests/channels/pjsip/sdp_offer_answer For example this one sets up a call, and also checks the 200 OK from Asterisk: https://github.com/asterisk/testsuite/tree/master/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/single-media-stream/audio/basic With the SIPp scenario being this: https://github.com/asterisk/testsuite/blob/master/tests/channels/pjsip/sdp_offer_answer/incoming/nominal/single-media-stream/audio/basic/sipp/uac-all-codecs.xml You'd need to add a re-INVITE in but there are also examples of that: https://github.com/asterisk/testsuite/blob/master/tests/channels/pjsip/connected_line/connected_line_allow/sipp/alice.xml#L67 General flow for such a test I'd expect to be: 1. SIPp scenario sends INVITE to Asterisk 2. Asterisk calls Answer() 3. Asterisk calls Wait(36) 4. SIPp scenario sends re-INVITE to Asterisk 5. Asterisk responds to re-INVITE 6. SIPp scenario checks 200 OK 7. SIPp sends BYE to Asterisk 8. Test automatically ends Asterisk would also have to have a configured endpoint with your new option. Cheers, -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com<http://www.sangoma.com> and www.asterisk.org<http://www.asterisk.org>
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