<div dir="ltr">Thanks, Nick. I'm having a read through it now.<div><br></div><div>I fixed my initial problem, though, and it was a pretty silly mistake: I had OpenSIPS and my SIP client (Jitsi) both using the same port. Oops! I switched Jitsi over to ports 55060/55061 and then things started working.</div>
<div><br></div><div>Working, at least with UDP. I need to get TCP working now.</div><div><br></div><div>The default configuration file has TCP disabled. Searching for information on how to set up TCP is quite difficult. The PDF you linked to has a few tips, luckily. But not enough to get it working.</div>
<div><br></div><div>Here's a portion of my config file:</div><div><br></div><div><div>#listen=udp:<a href="http://127.0.0.1:5060">127.0.0.1:5060</a> # CUSTOMIZE ME</div><div>listen=udp:<a href="http://172.16.23.79:5060">172.16.23.79:5060</a> # CUSTOMIZE ME</div>
<div>listen=tcp:<a href="http://172.16.23.79:5060">172.16.23.79:5060</a> # CUSTOMIZE ME</div><div>#disable_tcp=yes<br></div><div>disable_tcp=no</div><div>tcp_children=4</div></div><div><br></div><div style>When I try to connect with Jitsi, or with telnet, and sniff the traffic with Wireshark, I can see a SYN sent from Jitsi->OpenSIPS and a RST/ACK returned, but that's it. I don't think the TCP connection is getting properly established, and Jitsi bails out saying it can't create a connection.</div>
<div style><br></div><div style>Jitsi does successfully connect to OpenSIPS if I set the proxy transport setting to UDP in Jitsi.</div><div style><br></div><div style>What do I need to do to take OpenSIPS, from out of the box, and get it to accept REGISTERs/INVITEs on TCP and not UDP?</div>
<div style><br></div><div style>-PKCK</div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Feb 8, 2013 at 4:49 PM, Nick Khamis <span dir="ltr"><<a href="mailto:symack@gmail.com" target="_blank">symack@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Pink,<br>
<br>
A great resource that can get you up and going in at most a week can<br>
be found here:<br>
<br>
<br>
<a href="http://www.google.com/url?q=http://www.h6315.com/ast_book/Building%2520Telephony%2520Systems%2520with%2520OpenSIPS%25201.6.pdf&sa=U&ei=AJ0VUdahEvG10AGl4IDIDA&ved=0CB8QFjAA&sig2=q5gYZ0W2OeVVhS-YaZ0xjg&usg=AFQjCNFrbF0lgZG6kcTaZwI5uu-azQSx-w" target="_blank">http://www.google.com/url?q=http://www.h6315.com/ast_book/Building%2520Telephony%2520Systems%2520with%2520OpenSIPS%25201.6.pdf&sa=U&ei=AJ0VUdahEvG10AGl4IDIDA&ved=0CB8QFjAA&sig2=q5gYZ0W2OeVVhS-YaZ0xjg&usg=AFQjCNFrbF0lgZG6kcTaZwI5uu-azQSx-w</a><br>
<br>
Happy Routing!<br>
<br>
Nick.<br>
<div><div class="h5"><br>
On 2/8/13, Bogdan-Andrei Iancu <<a href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>> wrote:<br>
> Hi,<br>
><br>
> On 02/08/2013 12:17 AM, Pink Cupcake wrote:<br>
>> Hi Bogdan-Andrei,<br>
>><br>
>> Your response really doesn't help me or deal with the questions I laid<br>
>> out in my original post. Perhaps I was not being clear. Let me start<br>
>> over.<br>
>><br>
> Well, my answer was related to the error logs you posted :).<br>
><br>
>><br>
>><br>
>> I need to implement an automated testing scenario on an OS X build<br>
>> machine.<br>
>><br>
>> The test requires two different SIP UAs -- which are both running<br>
>> locally on the same machine -- to successfully engage in a SIP<br>
>> call. In order for two UAs to talk to one another, they need to be<br>
>> registered with a SIP server.<br>
>><br>
>> I am trying to determine if OpenSIPS can be used as the server<br>
>> component in my testing scenario.<br>
>><br>
>> Since this is an integration test and doesn't require any state being<br>
>> retained, the test steps look like this: (1) bring up an OpenSIPS<br>
>> server in userspace, (2) have the two SIP UA clients register with<br>
>> that server, as simply as possible, (3) have the two SIP UA clients<br>
>> engage in and then end a SIP call, (4) stop the SIP UA clients, (5)<br>
>> shut down the server.<br>
>><br>
>> If anything is unclear from the above, please reply back.<br>
> Clear and no issues here,<br>
><br>
>><br>
>><br>
>><br>
>> Please can someone answer the following:<br>
>><br>
>> If I can run OpenSIPS in userspace, I would also not like to have it<br>
>> "installed" on the build machine. I used the "prefix" parameter to<br>
>> `make install` into a separate directory and I am attempting to run<br>
>> OpenSIPS from that directory.<br>
>> It looks like I can run OpenSIPS in userspace. Is that correct?<br>
> true<br>
>><br>
>> From the documentation is looks like OpenSIPS does not use a database<br>
>> by default, and keeps everything in memory. Is that correct?<br>
> true<br>
>><br>
>> Can any SIP UA client "REGISTER" with OpenSIPS when it is launched in<br>
>> the "default" mode? If so, is there any special way the clients should<br>
>> send the request?<br>
> if using the opensips default script, no authentication will be required<br>
> - but you need to use in the REGISTER RURI the IP of the server (so that<br>
> opensips will consider them be handled locally).<br>
>><br>
>> If it is necessary for OpenSIPS to be run with a database in order to<br>
>> allow clients to register? If so, will the db_text module suffice? If<br>
>> so, how do I perform this configuration (given my testing scenario)?<br>
> no, no need for DB - by default, in cfg, the usrloc module comes with no<br>
> DB support.<br>
><br>
> Regards,<br>
> Bogdan<br>
>><br>
>><br>
>> PKCK<br>
>><br>
>><br>
>><br>
>> On Thu, Feb 7, 2013 at 5:13 AM, Bogdan-Andrei Iancu<br>
</div></div><div><div class="h5">>> <<a href="mailto:bogdan@opensips.org">bogdan@opensips.org</a> <mailto:<a href="mailto:bogdan@opensips.org">bogdan@opensips.org</a>>> wrote:<br>
>><br>
>> Hi,<br>
>><br>
>> Without a trace I cannot tell for sure, but I suspect your clients<br>
>> send several REGISTER requests without increasing the CSEQ no<br>
>> (which is mandatory) - this is the meaning of the error you get.<br>
>><br>
>> So, to be sure, make a network capture with the sip traffic<br>
>> (ngrep) and see what are the replies from opensips.<br>
>><br>
>> Regards,<br>
>><br>
>> Bogdan-Andrei Iancu<br>
>> OpenSIPS Founder and Developer<br>
>> <a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a><br>
>><br>
>><br>
>> On 02/07/2013 12:01 AM, Pink Cupcake wrote:<br>
>>> Hello,<br>
>>><br>
>>> I'm investigating the suitability of OpenSIPS for use in a new<br>
>>> system we are designing. Not only for use in a production<br>
>>> environment, but also how it can be used to facilitate automated<br>
>>> integration tests.<br>
>>><br>
>>> I have a automated testing scenario where I need to have two SIP<br>
>>> UAs that need to have a SIP session. What I would like to do is<br>
>>> bring up a SIP server (in userspace) before the integration test<br>
>>> starts, and bring it down after the integration test ends<br>
>>> (fails/succeeds). The automated test will run on OS X.<br>
>>><br>
>>> I downloaded OpenSIPS and built it on my iMac without any major<br>
>>> problems. I am able to run it in userspace simply by calling it<br>
>>> from the command line like `/sbin/opensips -D -f<br>
>>> /path/to/opensips.cfg`.<br>
>>><br>
>>> In section D of the INSTALL file, "opensips with Persistent Data<br>
>>> Storage", it says:<br>
>>><br>
>>> "The default configuration is very simple and features many<br>
>>> simplifications.<br>
>>> In particular, it does not authenticate users and loses User<br>
>>> Location database<br>
>>> on reboot. To provide persistence, keep user credentials and<br>
>>> remember users'<br>
>>> locations across reboots, opensips can be configured to use<br>
>>> MySQL. Before you<br>
>>> proceed, you need to make sure MySQL is installed on your box."<br>
>>><br>
>>> This sounds ideal to me; I don't need any real kind of account<br>
>>> management or authentication. I would like OpenSIPS to start,<br>
>>> accept whatever REGISTER/INVITE from my two UAs, and then stop<br>
>>> after I'm done. I would prefer not to require any database and<br>
>>> keep it all in-memory, so there's nothing to clean up before or<br>
>>> after the test (and no other dependencies to clean up before and<br>
>>> after, e.g. MySQL databases).<br>
>>><br>
>>> However, I can't seem to connect a SIP UA client to OpenSIPS when<br>
>>> it's started up like this. I am trying to connect with Jitsi, a<br>
>>> Mac client, as well as the ipjsua test app that ships with the<br>
>>> pjsip C library. (I am able to connect both of those to the<br>
</div></div>>>> <a href="http://sip2sip.info" target="_blank">sip2sip.info</a> <<a href="http://sip2sip.info" target="_blank">http://sip2sip.info</a>> service, so I know they are<br>
<div class="im">>>> both functional.)<br>
>>><br>
>>> With Jitsi, I set up a SIP account with Advanced settings<br>
>>> (username: test1, password: test1, display name: test1,<br>
>>> registrar: 127.0.0.1, port: 5060, manual proxy configuration,<br>
>>> proxy: 127.0.0.1, port: 5060).<br>
>>><br>
>>> Log output from opensips in Console.app looks like this:<br>
>>><br>
>>> 13-02-06 1:48:21.934 PM opensips: WARNING:core:warn: warning in<br>
>>> config file /path/to/opensips-with-local-changes.cfg, line 50,<br>
>>> column 13-16: tls support not compiled in<br>
>>> 13-02-06 1:48:22.010 PM opensips: WARNING:core:main: no fork mode<br>
>>> 13-02-06 1:48:22.011 PM opensips: NOTICE:core:main: version:<br>
>>> opensips 1.8.2-notls (x86_64/darwin)<br>
>>> 13-02-06 1:48:22.013 PM opensips: NOTICE:signaling:mod_init:<br>
>>> initializing module ...<br>
>>> 13-02-06 1:50:58.328 PM opensips:<br>
>>> ERROR:registrar:update_contacts: invalid cseq for aor <test1><br>
>>> 13-02-06 1:51:02.335 PM opensips:<br>
>>> ERROR:registrar:update_contacts: invalid cseq for aor <test1><br>
>>> 13-02-06 1:51:06.342 PM opensips:<br>
>>> ERROR:registrar:update_contacts: invalid cseq for aor <test1><br>
>>> ...<br>
>>><br>
>>> With ipjsua/pjsip, I use the following configuration switches:<br>
>>><br>
</div>>>> --id <a href="mailto:sip%3Atest1@127.0.0.1">sip:test1@127.0.0.1</a> <mailto:<a href="mailto:sip%253Atest1@127.0.0.1">sip%3Atest1@127.0.0.1</a>><br>
<div class="im">>>> --registrar sip:127.0.0.1<br>
>>> --realm *<br>
>>> --username test1<br>
>>> --password test1<br>
>>> --nameserver 127.0.0.1<br>
>>> --outbound sip:127.0.0.1<br>
>>><br>
>>> Log output in Console.app looks the same as with Jitsi except for<br>
>>> the "invalid cseq" lines:<br>
>>><br>
>>> 13-02-06 1:56:39.004 PM opensips:<br>
>>> ERROR:registrar:update_contacts: invalid cseq for aor <><br>
>>><br>
>>><br>
>>> What do I need to do to run OpenSIPS in userspace, have it accept<br>
>>> connections from my two SIP UAs, allow them to call each other,<br>
>>> and do it all without requiring a database running?<br>
>>><br>
>>> Do I absolutely require a database? If so, can someone explain<br>
>>> how to configure the db_text module to work for my testing scenario?<br>
>>><br>
>>> Thanks!<br>
>>><br>
>>><br>
>>> _______________________________________________<br>
>>> Users mailing list<br>
</div>>>> <a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a> <mailto:<a href="mailto:Users@lists.opensips.org">Users@lists.opensips.org</a>><br>
>>> <a href="http://lists.opensips.org/cgi-bin/mailman/listinfo/users" target="_blank">http://lists.opensips.org/cgi-bin/mailman/listinfo/users</a><br>
>><br>
>><br>
><br>
</blockquote></div><br></div>