Hey all,
I am in the process of setting up an openser machine for sip software
testsing to replace our ser machine. This decision came about due to ser
not supporting TLS in a readily available form and due to the seeming
stagnation of the ser project. I recently downloaded the 1.0.1-tls
source to compile. While in the process of duplicating our
configuration, I found that openser was complaining that there was a
mismatch between the module and the core, stating that the module was
1.0.0 and the core was 1.0.1-tls. Am I missing something in the "make
modules" command (make exclude_modules="" modules)? Do I need to add
something to the make file for pa.so to make it compile for the
appropriate version? Any help would be appreciated.
Justin Pearce
Price Video Productions
Justin(a)PriceVideo.com <mailto:Justin@PriceVideo.com>
Im currently learning OpenSer and have accomplished the following:
Installed: Centos 4.2
OpenSer: V1.0.1
I have 2 x-lite client registering and calling each other.
Question: How do I proxy on SIP requests to a 3rd part SIP Provider. eg.
Sipgate
xlite ---> Ser ---> Sipgate
Im just getting confused on how to configure the next hop routing?
Currently as expected im getting a 404 Not Found unless I call one of the
x-lite phones. I want to be able to terminate out to a SIP-to-PSTN provider.
Regards
Rob.
Hi guys, I'm having a bit of difficulty understanding when to use nathelper
and rtpproxy. If you wouldn't mind fielding my questions below, I'd really
appreciate it.
In what situations would you use:
-nathelper rather than rtpproxy?
-rtpproxy rather than nathelper?
-both nathelper and rtpproxy?
Thanks!
-- Nick
e: nick.hoffman(a)altcall.com
p: +61 7 5591 3588
f: +61 7 5591 6588
If you receive this email by mistake, please notify us and do not make any
use of the email. We do not waive any privilege, confidentiality or
copyright associated with it.
Add
if (!method=="REGISTER") record_route();
Somewhere in ur route (before relay but after loose_route)
do u have loose_route anyway ?
________________________________
From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On
Behalf Of Script Head
Sent: Thursday, March 09, 2006 6:44 PM
To: users(a)openser.org
Subject: Re: [Users] forcing rtpproxy on a call
Now that my rtpproxy actually passes traffic I stumbled on another
problem. When the called party hangs up the call (asterisk command
Hangup()) the soft phone remains connected. Yet, when I click the Hangup
button on the softphone, SER receives BYE messages.
On 3/9/06, Vitaly Nikolaev <vnikolaev(a)intermedia.net> wrote:
Looks like forward includes relay in it. And by putting force_rtpproxy
AFTER forward you does not give it a chance :-) on_reply route is also
MUST be there.
________________________________
From: users-bounces(a)openser.org [mailto: users-bounces(a)openser.org
<mailto:users-bounces@openser.org> ] On Behalf Of Script Head
Sent: Thursday, March 09, 2006 12:59 PM
To: users(a)openser.org
Subject: Re: [Users] forcing rtpproxy on a call
Thank you guys, it's working now.
Apparently, rewritehostport("<ip>:<port>") works great with rptproxy
while forward does exactly that, forwards the call to the destination
bypassing the force_rtp_proxy request. This should be documented
somewhere.
ScriptHead
On 3/9/06, Vitaly Nikolaev <vnikolaev(a)intermedia.net> wrote:
1. I never used forward, see my example, I do not know if it
actually relay call or not
2. if you do not have NAT between client and server you do not need
force_rport, and try to avoid any nat_uac_test, etc if you are actually
working on private ips without nat
3. you MUST enable proxy also for reply
route[x] {
.....
force_rtp_proxy();
t_on_reply("1");
rewritehostport("x.x.x.x:5060");
if (!t_relay()) {
sl_reply_error();
};
}
onreply_route[1] {
if (!(status=~"183" || status=~"200"))
break;
force_rtp_proxy("");
}
________________________________
From: users-bounces(a)openser.org [mailto: users-bounces(a)openser.org
<mailto:users-bounces@openser.org> ] On Behalf Of Script Head
Sent: Wednesday, March 08, 2006 6:29 PM
To: users(a)openser.org
Subject: [Users] forcing rtpproxy on a call
Hello everyone,
I am trying to debug why my rtpproxy isn't working. I have the following
setup, on my LAN.
softphone (192.168.1.100) -> openser/rtpproxy ( 192.168.1.10
<http://192.168.1.10> ) -> asterisk (192.168.1.12)
The rtpproxy is running and I see commands flying thru it.
the following route works
if(method=="INVITE") {
if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") {
forward(192.168.1.12,5060);
};
}
when I replace it with this route
if(method=="INVITE") {
if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") {
forward(192.168.1.12,5060);
};
force_rport();
force_rtp_proxy();
}
I get dead air while asterisk logs show that my test message is playing.
How should I proceed to debug this?
ScriptHead
All,
Been running 1.0.1 since last night and started noticing these errors,
can anyone indicate the seriousness of this error? It doesn't appear to
be affecting call flow.
BUG:tm:run_failure_handlers: stack level is 1!!
Looks like forward includes relay in it. And by putting force_rtpproxy
AFTER forward you does not give it a chance :-) on_reply route is also
MUST be there.
________________________________
From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On
Behalf Of Script Head
Sent: Thursday, March 09, 2006 12:59 PM
To: users(a)openser.org
Subject: Re: [Users] forcing rtpproxy on a call
Thank you guys, it's working now.
Apparently, rewritehostport("<ip>:<port>") works great with rptproxy
while forward does exactly that, forwards the call to the destination
bypassing the force_rtp_proxy request. This should be documented
somewhere.
ScriptHead
On 3/9/06, Vitaly Nikolaev <vnikolaev(a)intermedia.net> wrote:
1. I never used forward, see my example, I do not know if it
actually relay call or not
2. if you do not have NAT between client and server you do not need
force_rport, and try to avoid any nat_uac_test, etc if you are actually
working on private ips without nat
3. you MUST enable proxy also for reply
route[x] {
.....
force_rtp_proxy();
t_on_reply("1");
rewritehostport("x.x.x.x:5060");
if (!t_relay()) {
sl_reply_error();
};
}
onreply_route[1] {
if (!(status=~"183" || status=~"200"))
break;
force_rtp_proxy("");
}
________________________________
From: users-bounces(a)openser.org [mailto: users-bounces(a)openser.org
<mailto:users-bounces@openser.org> ] On Behalf Of Script Head
Sent: Wednesday, March 08, 2006 6:29 PM
To: users(a)openser.org
Subject: [Users] forcing rtpproxy on a call
Hello everyone,
I am trying to debug why my rtpproxy isn't working. I have the following
setup, on my LAN.
softphone (192.168.1.100) -> openser/rtpproxy ( 192.168.1.10
<http://192.168.1.10> ) -> asterisk (192.168.1.12)
The rtpproxy is running and I see commands flying thru it.
the following route works
if(method=="INVITE") {
if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") {
forward(192.168.1.12,5060);
};
}
when I replace it with this route
if(method=="INVITE") {
if(uri=~"^sip:[0-9]{6}1[0-9]*{10}@") {
forward(192.168.1.12,5060);
};
force_rport();
force_rtp_proxy();
}
I get dead air while asterisk logs show that my test message is playing.
How should I proceed to debug this?
ScriptHead
Hi!
I've been running some performance tests on my OpenSER box and while
registering large number of users I faced some very strange problems. At
least they seem strange to me, but I hope someone on this list would be able
to shed some light on this.
So, I use Openser 1.0.1 and I load test it using SIPp. I've done tests
creating about 1000 registrations per second registering 300k-450k different
users. Register rate is constat and for each register different user is
used. What is strange to me that just about when 300k users have registered
the CPU of the PC running the proxy hits 100% usage and messages start
dropping. I made a nice image to illustrate my problem using Excel. I
attached it to this mail. On this test I registered 300k different users at
1000 RPS and after registering those users the registering starts again from
the beginning. As you can see from it the CPU usage rises constantly and
drops immediately after I've registered 300k users and the registering
starts again from the start. So then each user is being registered all over
again, but still the CPU load grows. Does anyone have explanation for this?
Why does the CPU usage grow based on the number of users registered? And why
does it drop for a while when registering starts all over again? I've tried
different usrloc modes, but there seems to be no difference. This one was
done with having usrloc only in the memory.
Regards,
Teemu
--
Teemu Harju
http://www.teemuharju.net
Hi!
I posted about this same subject to openser mailing list already, but since
I haven't had response from there I'll try here. I guess the usrloc module
and the core functionality is quite the same with ser and openser.
I've been running some performance tests on my OpenSER box and while
registering large number of users I faced some very strange problems. At
least they seem strange to me, but I hope someone on this list would be able
to shed some light on this.
So, I use Openser 1.0.1 and I load test it using SIPp. I've done tests
creating about 1000 registrations per second registering 300k-450k different
users. Register rate is constat and for each register different user is
used. What is strange to me that just about when 300k users have registered
the CPU of the PC running the proxy hits 100% usage and messages start
dropping. I made a nice image to illustrate my problem using Excel. Check it
at http://www.teemuharju.net/wp-images/register_300k_users_at_1000rps.png.
On this test I registered 300k different users at 1000 RPS and after
registering those users the registering starts again from the beginning. As
you can see from it the CPU usage rises constantly and drops immediately
after I've registered 300k users and the registering starts again from the
start. So then each user is being registered all over again, but still the
CPU load grows. Does anyone have explanation for this? Why does the CPU
usage grow based on the number of users registered? And why does it drop for
a while when registering starts all over again? I've tried different usrloc
modes, but there seems to be no difference. This one was done with having
usrloc only in the memory. Shouldn't the CPU usage at constant register
rate, be somewhat constant also? Has anyone else noticed the same problem?
Could this be somehow explained or do I have something wrong with my system?
Regards,
Teemu
--
Teemu Harju
http://www.teemuharju.net
Hi Friend,
Can I ask you to suggest me for any of the Java based Softphone that you are using it in the real time environment.
Point me to any of the stable releases. I just came to know the "SipCommunicator" a java based one, but it is not stable enough.
Thanks ya,
Se.
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