Hi,
On 02/08/2013 12:17 AM, Pink Cupcake wrote:
Hi Bogdan-Andrei,
Your response really doesn't help me or deal with the questions I laid
out in my original post. Perhaps I was not being clear. Let me start over.
Well, my answer was related to the error logs you posted :).
I need to implement an automated testing scenario on an OS X build
machine.
The test requires two different SIP UAs -- which are both running
locally on the same machine -- to successfully engage in a SIP
call. In order for two UAs to talk to one another, they need to be
registered with a SIP server.
I am trying to determine if OpenSIPS can be used as the server
component in my testing scenario.
Since this is an integration test and doesn't require any state being
retained, the test steps look like this: (1) bring up an OpenSIPS
server in userspace, (2) have the two SIP UA clients register with
that server, as simply as possible, (3) have the two SIP UA clients
engage in and then end a SIP call, (4) stop the SIP UA clients, (5)
shut down the server.
If anything is unclear from the above, please reply back.
Clear and no issues here,
Please can someone answer the following:
If I can run OpenSIPS in userspace, I would also not like to have it
"installed" on the build machine. I used the "prefix" parameter to
`make install` into a separate directory and I am attempting to run
OpenSIPS from that directory.
It looks like I can run OpenSIPS in userspace. Is that correct?
true
From the documentation is looks like OpenSIPS does not use a database
by default, and keeps everything in memory. Is that correct?
true
Can any SIP UA client "REGISTER" with OpenSIPS when it is launched in
the "default" mode? If so, is there any special way the clients should
send the request?
if using the opensips default script, no authentication will be required
- but you need to use in the REGISTER RURI the IP of the server (so that
opensips will consider them be handled locally).
If it is necessary for OpenSIPS to be run with a database in order to
allow clients to register? If so, will the db_text module suffice? If
so, how do I perform this configuration (given my testing scenario)?
no, no need for DB - by default, in cfg, the usrloc module comes with no
DB support.
Regards,
Bogdan
PKCK
On Thu, Feb 7, 2013 at 5:13 AM, Bogdan-Andrei Iancu
<[email protected] <mailto:[email protected]>> wrote:
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:[email protected] <mailto:sip%[email protected]>
--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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