[OpenSIPS-Users] Automated Testing Scenario with OpenSIPS

[Pink Cupcake]

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 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
--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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