On Thu, Sep 29, 2022 at 9:15 AM Henning Westerholt <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 and www.asterisk.org
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