[OpenSIPS-Users] Automated Testing Scenario with OpenSIPS

[Bogdan-Andrei Iancu]

Hi,

Without a trace I cannot tell for sure, but I suspect your clients send 
several REGISTER requests without increasing the CSEQ no (which is 
mandatory) - this is the meaning of the error you get.

So, to be sure, make a network capture with the sip traffic (ngrep) and 
see what are the replies from opensips.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 02/07/2013 12:01 AM, Pink Cupcake wrote:
> Hello,
>
> I'm investigating the suitability of OpenSIPS for use in a new system 
> we are designing. Not only for use in a production environment, but 
> also how it can be used to facilitate automated integration tests.
>
> I have a automated testing scenario where I need to have two SIP UAs 
> that need to have a SIP session. What I would like to do is bring up a 
> SIP server (in userspace) before the integration test starts, and 
> bring it down after the integration test ends (fails/succeeds). The 
> automated test will run on OS X.
>
> I downloaded OpenSIPS and built it on my iMac without any major 
> problems. I am able to run it in userspace simply by calling it from 
> the command line like `/sbin/opensips -D -f /path/to/opensips.cfg`.
>
> In section D of the INSTALL file, "opensips with Persistent Data 
> Storage", it says:
>
> "The default configuration is very simple and features many 
> simplifications.
> In particular, it does not authenticate users and loses User Location 
> database
> on reboot. To provide persistence, keep user credentials and remember 
> users'
> locations across reboots, opensips can be configured to use MySQL. 
> Before you
> proceed, you need to make sure MySQL is installed on your box."
>
> This sounds ideal to me; I don't need any real kind of account 
> management or authentication. I would like OpenSIPS to start, accept 
> whatever REGISTER/INVITE from my two UAs, and then stop after I'm 
> done. I would prefer not to require any database and keep it all 
> in-memory, so there's nothing to clean up before or after the test 
> (and no other dependencies to clean up before and after, e.g. MySQL 
> databases).
>
> However, I can't seem to connect a SIP UA client to OpenSIPS when it's 
> started up like this. I am trying to connect with Jitsi, a Mac client, 
> as well as the ipjsua test app that ships with the pjsip C library. (I 
> am able to connect both of those to the sip2sip.info 
> <http://sip2sip.info> service, so I know they are both functional.)
>
> With Jitsi, I set up a SIP account with Advanced settings (username: 
> test1, password: test1, display name: test1, registrar: 127.0.0.1, 
> port: 5060, manual proxy configuration, proxy: 127.0.0.1, port: 5060).
>
> Log output from opensips in Console.app looks like this:
>
> 13-02-06 1:48:21.934 PM opensips: WARNING:core:warn: warning in config 
> file /path/to/opensips-with-local-changes.cfg, line 50, column 13-16: 
> tls support not compiled in
> 13-02-06 1:48:22.010 PM opensips: WARNING:core:main: no fork mode
> 13-02-06 1:48:22.011 PM opensips: NOTICE:core:main: version: opensips 
> 1.8.2-notls (x86_64/darwin)
> 13-02-06 1:48:22.013 PM opensips: NOTICE:signaling:mod_init: 
> initializing module ...
> 13-02-06 1:50:58.328 PM opensips: ERROR:registrar:update_contacts: 
> invalid cseq for aor <test1>
> 13-02-06 1:51:02.335 PM opensips: ERROR:registrar:update_contacts: 
> invalid cseq for aor <test1>
> 13-02-06 1:51:06.342 PM opensips: ERROR:registrar:update_contacts: 
> invalid cseq for aor <test1>
> ...
>
> With ipjsua/pjsip, I use the following configuration switches:
>
> --id sip:test1 at 127.0.0.1 <mailto:sip%3Atest1 at 127.0.0.1>
> --registrar sip:127.0.0.1
> --realm *
> --username test1
> --password test1
> --nameserver 127.0.0.1
> --outbound sip:127.0.0.1
>
> Log output in Console.app looks the same as with Jitsi except for the 
> "invalid cseq" lines:
>
> 13-02-06 1:56:39.004 PM opensips: ERROR:registrar:update_contacts: 
> invalid cseq for aor <>
>
>
> What do I need to do to run OpenSIPS in userspace, have it accept 
> connections from my two SIP UAs, allow them to call each other, and do 
> it all without requiring a database running?
>
> Do I absolutely require a database? If so, can someone explain how to 
> configure the db_text module to work for my testing scenario?
>
> Thanks!
>
>
> _______________________________________________
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> Users at lists.opensips.org
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