[Sipp-users] Problem using OpenSIPS with SIPp for testing
I am stuck with a problem while trying to use OpenSIPS and SIPp. I hope
someone can help me out.
I have OpenSIPS runnin on 10.1.10.88
I am using the following cfg file
debug=3
*fork=no
log_stderror=yes
listen=10.1.10.88
port=5060
children=4
dns=yes
rev_dns=yes
#fifo= /tmp/ser_fifo
loadmodule /usr/src/opensips-1.4.4-tls/modules/sl/sl.so
loadmodule /usr/src/opensips-1.4.4-tls/modules/tm/tm.so
loadmodule /usr/src/opensips-1.4.4-tls/modules/rr/rr.so
loadmodule /usr/src/opensips-1.4.4-tls/modules/maxfwd/maxfwd.so
loadmodule /usr/src/opensips-1.4.4-tls/modules/usrloc/usrloc.so
loadmodule /usr/src/opensips-1.4.4-tls/modules/registrar/registrar.so
modparam(usrloc, db_mode, 0)
modparam(rr, enable_full_lr, 1)
route {
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483, Too Many Hops);
return;
};
if (msg:len max_len) {
sl_send_reply(513, Message Overflow);
return;
};
if (method!=REGISTER) {
record_route();
};
if (loose_route()) {
route(1);
return;
};
if (uri!=myself) {
route(1);
return;
};
if (method==ACK) {
route(1);
return;
}
if (method==REGISTER) {
route(2);
return;
};
if (method==INVITE) {
add_local_rport();
force_rport();
forward();
t_relay();
return;
};
lookup(aliases);
if (uri!=myself) {
route(1);
return;
};
if (!lookup(location)) {
sl_send_reply(404, User Not Found);
return;
};
route(1);
}
route[1]{
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
if (!save(location)) {
sl_reply_error();
};
}
*
I have used the following cmds on Three diff Desktops
*opensips -f /usr/src/opensips-1.4.4-tls/examples/new.cfg -l 10.1.10.88:5060
*
*sipp -sn -uac -r 20 -rp 200 -i 10.1.10.87 -p 5062 -rsa 10.1.10.88:5060
10.1.10.86:5061*
*sipp -sn uas -i 10.1.10.86 -p 5061 -rsa 10.1.10.88:5060*
When the UAC starts sending the request they are completely absorbed by
OpenSIPS and not at all forwarded to UAS. As a result the Count of INVITE
and 100 are going up in SIPP (UAC) but then there is nothing happening after
that. I mean there is no response from UAS. The 180, BYE on UAC count to
zero till very end and there no increament in the messages on UAS list.
All the calls are shown to be paused on UAC.
I tried lot of debugging with the cfg file and have changed lot of things
there (compared to the first one I wrote). I tried swapping the role of UAS
and UAC. The problem persists with UAC (it seems to be machine independent,
and not partial towards 10.1.10.87 :-) ).
I tried running OpenSIPS and SIPp all on local host(and used IP 127.0.0.1 in
all commands), and still the problem was there. So network can not be held
responsible for this problem.
I have tried the /opensips/examples/logging.cfg file (with some
modification) and confirmed that the INVITE messages are successfully
recieved and logged by OpenSIPS (at 10.1.10.88).
I even tried running OpenSIPS on the machine running UAS (10.1.10.86)
scenario and there I tried to log (with a similar logging.cfg file) incoming
messages but I failed to find any INVITE message relayed by the OpenSIPS on
10.1.10.88.
Sameer Kumar
B.Tech, Computer Engineering
Institute of Technology,
Nirma University,
Ahmedabad, Gujarat
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Re: [Sipp-users] Problem using OpenSIPS with SIPp for testing
On Fri, Mar 13, 2009 at 7:05 PM, SAMEER KUMAR sameer.kasi2...@gmail.com wrote:
if (method==INVITE) {
add_local_rport();
force_rport();
forward();
t_relay();
Hello,
why are you calling both forward (stateless) and t_relay (stateful)
when processing the INVITE?
Maybe this is causing opensips to misbehave.
regards,
takeshi
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Re: [Sipp-users] A more robust SIPp
Unfortunately, SIPp uses a built-in XML parser using strstr. Charles Kirwan, David (David) dkir...@avaya.com 03/13/2009 06:29 AM To sipp-users@lists.sourceforge.net cc Subject [Sipp-users] A more robust SIPp Hi, While writting a SIPp script, I omitted a forward slash. The script ran, but didn't send the ACK, which was a few lines below the line of XML that was missing the forward slash. I'm sure SIPp has a fully fledged XML parser onboard, probably IBM's xerces parser, so why can't SIPp tell you the XML is or isn't well-formed and valid before trying to run the script? Best regards, David -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
Re: [Sipp-users] Problem using OpenSIPS with SIPp for testing
On Fri, Mar 13, 2009 at 9:04 PM, SAMEER KUMAR sameer.kasi2...@gmail.com wrote:
Dear,
Thanks for such a prompt reply.
Earlier I was not even using a saperate processing for INVITE. I was
supposed to be processed by the fefault call to route[1] (please refer to
the cfg file). But as I noticed the problem I stated, I tought that the call
to t_relay in route[1] is not being processed (in route[1] in the if
condition check). So I made a explicit processing for INVITE and then used
t_relay. But when that too did not seemed to help I used stateless forward.
I had tried lot many different things (like adding via and record route
functions etc etc) and also tested other example cfg files. So be sure that
there is no problem with cfg file. The problem persists ireespective of all
that.
Hello Sameer,
my current installation is not compiled with tls and I also don't have
the load_sig_api required by the registrar module. So I cleared your
cfg from registrar functions and running it I can see it does work
(i'm using opensips head r. 5438). Here's the modified cfg file:
debug=3
fork=no
log_stderror=yes
listen=192.168.2.3
port=5080
children=4
dns=yes
rev_dns=yes
#fifo= /tmp/ser_fifo
mpath=/usr/local/lib/opensips/modules
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
modparam(usrloc, db_mode, 0)
modparam(rr, enable_full_lr, 1)
route {
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483, Too Many Hops);
return;
};
if (msg:len max_len) {
sl_send_reply(513, Message Overflow);
return;
};
if (method!=REGISTER) {
record_route();
};
if (loose_route()) {
route(1);
return;
};
if (uri!=myself) {
route(1);
return;
};
if (method==ACK) {
route(1);
return;
}
if (method==INVITE) {
add_local_rport();
force_rport();
forward();
t_relay();
return;
};
route(1);
}
route[1] {
if (!t_relay()) {
sl_reply_error();
};
}
Maybe this is something related to TLS. Have you tried testing with no TLS?
--
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powering Web 2.0 with engaging, cross-platform capabilities. Quickly and
easily build your RIAs with Flex Builder, the Eclipse(TM)based development
software that enables intelligent coding and step-through debugging.
Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com
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Re: [Sipp-users] Problem using OpenSIPS with SIPp for testing
Dear,
Yes, I had tried with disable tls option set and clear. But could not find
any difference in behavior. Though I will be trying again the new
suggestion of yours to use without registerar module. But that will be only
in the morning when I visit my lab.
Thanks alot for the prompt reply and taking interest.
Sameer Kumar
B.Tech, Computer Engineering
Institute of Technology,
Nirma University,
Ahmedabad, Gujarat
On Fri, Mar 13, 2009 at 6:21 PM, mayamatakeshi mayamatake...@gmail.comwrote:
On Fri, Mar 13, 2009 at 9:04 PM, SAMEER KUMAR sameer.kasi2...@gmail.com
wrote:
Dear,
Thanks for such a prompt reply.
Earlier I was not even using a saperate processing for INVITE. I was
supposed to be processed by the fefault call to route[1] (please refer to
the cfg file). But as I noticed the problem I stated, I tought that the
call
to t_relay in route[1] is not being processed (in route[1] in the if
condition check). So I made a explicit processing for INVITE and then
used
t_relay. But when that too did not seemed to help I used stateless
forward.
I had tried lot many different things (like adding via and record route
functions etc etc) and also tested other example cfg files. So be sure
that
there is no problem with cfg file. The problem persists ireespective of
all
that.
Hello Sameer,
my current installation is not compiled with tls and I also don't have
the load_sig_api required by the registrar module. So I cleared your
cfg from registrar functions and running it I can see it does work
(i'm using opensips head r. 5438). Here's the modified cfg file:
debug=3
fork=no
log_stderror=yes
listen=192.168.2.3
port=5080
children=4
dns=yes
rev_dns=yes
#fifo= /tmp/ser_fifo
mpath=/usr/local/lib/opensips/modules
loadmodule sl.so
loadmodule tm.so
loadmodule rr.so
loadmodule maxfwd.so
loadmodule usrloc.so
modparam(usrloc, db_mode, 0)
modparam(rr, enable_full_lr, 1)
route {
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483, Too Many Hops);
return;
};
if (msg:len max_len) {
sl_send_reply(513, Message Overflow);
return;
};
if (method!=REGISTER) {
record_route();
};
if (loose_route()) {
route(1);
return;
};
if (uri!=myself) {
route(1);
return;
};
if (method==ACK) {
route(1);
return;
}
if (method==INVITE) {
add_local_rport();
force_rport();
forward();
t_relay();
return;
};
route(1);
}
route[1] {
if (!t_relay()) {
sl_reply_error();
};
}
Maybe this is something related to TLS. Have you tried testing with no TLS?
--
Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are
powering Web 2.0 with engaging, cross-platform capabilities. Quickly and
easily build your RIAs with Flex Builder, the Eclipse(TM)based development
software that enables intelligent coding and step-through debugging.
Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com___
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[Sipp-users] asterisk/sipp rtp port bug?
After a lot of debugging with asterisk and sipp in pcap mode we found the following problem: If asterisk is configured to allow rtp ports greater than , it seems the last digit is trimmed away some how. An example of a tcpdump followed by an immediate netstat shows this problem: -192.168.13.198 is running sipp in uac_pcap mode. -192.168.13.150 is the asterisk server. -uac is sending packets to udp port 1830 -asterisk is listening on ports 18300 and 18301 If asterisk happens to choose a port below 1 (a port that is 4 digits long), everything goes well. This looks like a SIP parser bug in sipp somewhere, since normally we dont have this problem with other sip clients. The SIP protocol is text-based, so a parser bug makes sence, right? Could somebody look into the sourcecode if this is right? If it IS a bug, it would certainly explain a lot for many people i think ;) Edwin Eefting http://www.syn-3.nl -(paste of terminal output) [Syn-3] r...@darkstar.example.net /etc/asterisk# tcpdump -n -l src 192.168.13.198 and not port 22 or proto ICMP | head -20 ; netstat -nap| grep asterisk tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth0:0, link-type EN10MB (Ethernet), capture size 68 bytes 17:57:49.398169 IP 192.168.13.198.6016 192.168.13.150.1830: UDP, length 252 17:57:49.428113 IP 192.168.13.198.6016 192.168.13.150.1830: UDP, length 252 17:57:49.458880 IP 192.168.13.198.6016 192.168.13.150.1830: UDP, length 252 17:57:49.458939 IP 192.168.13.150 192.168.13.198: ICMP 192.168.13.150 udp port 1830 unreachable, length 288 17:57:49.488563 IP 192.168.13.198.6016 192.168.13.150.1830: UDP, length 252 17:57:49.517853 IP 192.168.13.198.6016 192.168.13.150.1830: UDP, length 252 21 packets captured 48 packets received by filter 0 packets dropped by kernel udp0 0 0.0.0.0:50600.0.0.0:* 18189/asterisk udp0 0 0.0.0.0:18300 0.0.0.0:* 18189/asterisk udp0 0 0.0.0.0:18301 0.0.0.0:* 18189/asterisk udp0 0 0.0.0.0:17278 0.0.0.0:* 18189/asterisk -- -- Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are powering Web 2.0 with engaging, cross-platform capabilities. Quickly and easily build your RIAs with Flex Builder, the Eclipse(TM)based development software that enables intelligent coding and step-through debugging. Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com ___ Sipp-users mailing list Sipp-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sipp-users
