once in a while kamailio 4.0 presence server becomes unresponsive, i.e.,
does not process any requests. below is bt full of a process that at
that time takes most of the cpu time. rls_notifier_processes is not
set, i.e., it defaults to 1.
does the bt give any clue why kamailio is unresponsive?
-- juha
(gdb) bt full
#0 0xb7703424 in __kernel_vsyscall ()
No symbol table info available.
#1 0xb765b32d in select () from /lib/i686/cmov/libc.so.6
No symbol table info available.
#2 0x0812ad00 in sleep_us (child_id=-1, desc=0xbfab3310 "RLS NOTIFIER 0", make_sock=1, f=0xb6b22dc0 <timer_send_notify>, param=0xb4db0720, uinterval=100000) at ut.h:520
tval = {tv_sec = 0, tv_usec = 64460}
#3 fork_basic_utimer (child_id=-1, desc=0xbfab3310 "RLS NOTIFIER 0", make_sock=1, f=0xb6b22dc0 <timer_send_notify>, param=0xb4db0720, uinterval=100000) at timer_proc.c:127
pid = <value optimized out>
ts = 4294966782
#4 0xb6b23b90 in child_init (rank=0) at rls.c:704
tmp = "RLS NOTIFIER 0\000\277"
i = 0
#5 0x080f5197 in init_mod_child (m=0xb71a3330, rank=0) at sr_module.c:893
No locals.
#6 0x080f5110 in init_mod_child (m=0xb71a3500, rank=0) at sr_module.c:890
No locals.
#7 0x080f5110 in init_mod_child (m=0xb71a37b0, rank=0) at sr_module.c:890
No locals.
#8 0x080f5110 in init_mod_child (m=0xb71a3c20, rank=0) at sr_module.c:890
No locals.
#9 0x080f5110 in init_mod_child (m=0xb71a4258, rank=0) at sr_module.c:890
No locals.
#10 0x080f5110 in init_mod_child (m=0xb71a4420, rank=0) at sr_module.c:890
No locals.
#11 0x08094dcd in main_loop () at main.c:1710
i = 0
pid = -514
si = 0x0
si_desc = "\001\000\000\000\220\065\253\277\000\000\000\000\220D\032\267\006\000\000\000\020\317\060\000\000\000\000\000\220D\032\267\001\000\000\000\330\067\005\b\320i\036\b\000\000\000\000\030\036Z\267\030\000\000\000\v\b\000\000\b\017۴\350\065\253\277\250\205\031\267\004\000\000\000\002\000\000\000\300\201\252\264\001\000\000\000\000\000\000\000\002\000\000\000|\353&\b\b\000\000\000\330\065\253\277\002\000\000\000h\353&\b\b\000\000\000\350\065\253\277\071\026\f\b"
nrprocs = 134560983
#12 0x08096f66 in main (argc=16, argv=0xbfab3724) at main.c:2546
cfg_stream = 0x8
c = <value optimized out>
r = -514
tmp = 0xbfab3f7d ""
tmp_len = 135830000
port = <value optimized out>
proto = <value optimized out>
ret = <value optimized out>
seed = 779380118
rfd = <value optimized out>
debug_save = 0
debug_flag = <value optimized out>
dont_fork_cnt = 8
n_lst = <value optimized out>
p = <value optimized out>
Hey all,
I have kamailio set up behind a nat(port restricted, with firewall rules to
allow sip transactions and allowing rtpproxy packet handling if needed) on
Amazon EC2. I can register and calls complete, however, the Caller(the one
initiating the transaction) receives no rtp media feed. I am running with
NAT enabled on kamailio and have rtpproxy installed listening on the public
IP. Kamailio and the rtpproxy are communicating(I have verified via the
kamailio debug logs). If I make a call between the exact same voip machines
directly via local IP on the same Nat(skipping kamailio), the calls
complete and they both receive feeds.
Both the Caller(party making the call) and the Callee(party receiving the
call) are behind a Port Restricted Nat.
This is a folder containing the debug output for one of these calls and the
kamailio.cfg settings
https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp=sha…
Quick FYI, the Caller Display Name and the Callee SIP UserName are the same
string. However, in my understanding about sip, the display name means
pretty much nothing and is just a human readable string that does not
effect packet transport. If I am wrong and should test with a different
display name, let me know.
Thank you for the assistance,
ben
Hi All,
root@sip-router3-ve206:/etc/kamailio# kamailio -V
version: kamailio 4.1.0 (x86_64/linux) 350d2e
Doing some testing, can't seem to get rtpproxy to not segfault. I've
loaded version from deb-squeeze pkg, from source
http://b2bua.org/chrome/site/rtpproxy-1.2.1.tar.gz and from git://
sippy.git.sourceforge.net/gitroot/sippy/rtpproxy (which should be the
latest) but all versions segfault as soon as a call sets up.
I've configured each versions control socket for both udp or unix, kamailio
starts and sees the rtpproxy fine with no errors, but when a call hits,
rtpproxy segfaults.
I've run rtpproxy in the forground with degug and get this response (IP's
washed x.x.x):
------------------
DBUG:handle_command: received command "7505_12 USIEc0,101
0adb2f8449b8c9026f993a0a7db9ab5d(a)x.x.x.76 x.x.x.76 23388 as33dd7c98;1"
INFO:handle_command: new session 0adb2f8449b8c9026f993a0a7db9ab5d(a)x.x.x.76,
tag as33dd7c98;1 requested, type strong
Segmentation fault
------------------
root@sip-router3-ve206:/etc/kamailio# more /etc/default/rtpproxy
# Defaults for rtpproxy
# The control socket.
CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock"
# To listen on an UDP socket, uncomment this line:
#CONTROL_SOCK="udp:127.0.0.1:7722"
# Additional options that are passed to the daemon.
EXTRA_OPTS="-l x.x.x.20"
kamailio.cfg:
loadmodule "rtpproxy.so"
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
inside route[]
add_path_received();
rtpproxy_manage("cwei");
record_route();
Any guidance on further identifying the issue?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
Hello,
Kamailio SIP Server v4.0.5 stable release is out.
This is a maintenance release of the stable branch 4.0, that includes
fixes since release of v4.0.4. There is no change to database schema or
configuration language structure that you have to do on installations of
v4.0.x. Deployments running previous v4.0.x versions are strongly
recommended to be upgraded to v4.0.5.
For more details about version 4.0.5 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2013/12/kamailio-v4-0-5-released/
Note that 4.0.x is previous stable release series, the latest one is now
4.1.x, with v4.1.0 released about two weeks ago.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hey all,
**I apologize if this is a duplicate, I received a bounce back on my first
email.
I have kamailio set up behind a nat(port restricted, with firewall rules to
allow sip transactions and allowing rtpproxy packet handling if needed) on
Amazon EC2. I can register and calls complete, however, the Caller(the one
initiating the transaction) receives no rtp media feed. I am running with
NAT enabled on kamailio and have rtpproxy installed listening on the public
IP. Kamailio and the rtpproxy are communicating(I have verified via the
kamailio debug logs). If I make a call between the exact same voip machines
directly via local IP on the same Nat(skipping kamailio), the calls
complete and they both receive feeds.
Both the Caller(party making the call) and the Callee(party receiving the
call) are behind a Port Restricted Nat.
This is a folder containing the debug output for one of these calls and the
kamailio.cfg settings
https://drive.google.com/folderview?id=0B9Foq0jDF8gLRlVNc001bTUtbFE&usp=sha…
Quick FYI, the Caller Display Name and the Callee SIP UserName are the same
string. However, in my understanding about sip, the display name means
pretty much nothing and is just a human readable string that does not
effect packet transport. If I am wrong and should test with a different
display name, let me know.
Thank you for the assistance,
ben
hello,
I’m trying to build two kamailio-3.3.4 server like this:
ua1 ----> kamailio1(k1)-----> kamailio2(k2) -----> ua2
ua1 (5978003 ) is register k1
ua2 (5001005) is register k2
When ua1 callout to ua2 sip process through k1 and k2
The problem is k2 server ACK not transfer to ua2 and k2 has received ack
signals from u1 through k1
I ngrep sip process for k2 and k2 /var/log/debug
see the attached File ngrep_k2server.txt, debug_k2
1.
First ACK uri have ua2 number sip:5001005@xxx.xxx.xxx.xxx(k2):5003;ob ,
is correct !!
2. Second ack not include 5001005 , again loop itself ; Route:
<sip:xxx.xxx.xxx.xxx(k2):5003;lr;did=763.941>
loose_route Next hop: 'sip:xxx.xxx.xxx.xxx(k2):5003;lr;did=763.941' is
loose router.
so next hop Not found in usrloc db 5001005 user and not transfer to
ua2
How to fix this problem, any suggestions ?
Dear All,
I am running a kamailio 4.0.3 server, it was working fine with the features
like- audio calling, video calling , SMS (but presence is not working). But
from last two days I’m facing problems with RTP packets streaming between
SIP clients (IMSDroid clients). Session establishment between two clients
works well but without audio and video streaming. And after sometime again
it works well. What is this wierd behaviour?
Following attachments are Audio calling cases of working and not working
(ngrep based), And i have found some errors in Syslog (attached) and my
config file for any reference.
What could be the problem? How can i solve the issue?
Anybody please help me.
Regards,
Hello,
I am considering to release a new version out of branch 4.0 (previous
stable branch) before Christmas holidays. Depending on the available
time, it could be this Friday or next week on Monday. If there are
patches to be backported, do them as soon as possible.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda