Hello,
After days of debugging, I'll need some help from the SER specialists to
troubleshoot this problem.
The SER version in service is: 0.9.9+cvs20090925
It's running on a Red Hat EL 5.3 server.
I currently have about 10.000 users registered.
I encounter several seconds (1 to 5) without replies from the SIP proxy
every 30 seconds.
Some requests are delayed, but most of them are not replied at all.
At the same time, all the 6 SER childs (forking is enabled) get about 100%
used CPU (from "top")
After some troubleshooting, I found that the rythm of those deny of service
was exactly the time configured as the "timer_interval" parameter
of "usrloc" module.
When changing the parameter, problem rythm changes as well.
What we did so far:
- Check DNS and MySQL performances
- Moved the number of childs from 6 to 9
- Changed the "timer_interval" ("usrloc" module) from 60 to 900
- Stopped the "natping" feature ("nathelper" module) as it impacted the
service as well.
Please let me know if the problem has already been observed and what would
be the best way to go deeper in the troubleshooting.
I need to solve that as soon as possible: some of our users higly suffer of
random REGISTER not beeing taken into account.
Sincerely,
Igor
I'm trying to figure out if there is some way to identify a call for
which an rtpproxy session has been set up when the call is cancelled.
I can't check the flag I set for the call, nor can I check the route
param I set (which is what I'm doing to figure out when to unforce the
media stream for a BYE).
If this is not possible, is there any danger in calling
unforce_rtp_proxy() for all calls, regardless of whether media proxying
has been set up for the call or not?
--
Joe Hart
Voice Systems Integrator
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web : http://www.evaristesys.com/
Hi there,
Is there a way to relay call to secure gateway or ITSP.
ie, invite being challange to provide authorization (username/password)
Could someone please give me some advice if this is possible.
Thanks,
Jkwan
Hi All,
I'm facing an issue with TCP connection from Proxy(kamailio) to a SIP server
on a load test, where kamailio is not processing the message fast enough and
which leads socket buffer getting filled up. I have increased the
tcp_rd_buf_size but that really didn't help.
According to my understanding the each TCP connection is assigned to a TCP
children, and TCP children will read until it there is no data for few
seconds and return the connection to tcp main process. In this case the TCP
connection is to the SIP server which always have some data due to periodic
SIP ping ,registration refresh and calls etc. So i believe this connection
will be hold always by one children and since that children is processing
the message sequentially, it might not be possible to increase the
performance on this connection.
My question is, If my understanding is correct How do we increase the
performance for a particular TCP connection? Does Kamailio has some
mechanism to distribute the message from a single TCP connection
to different child, when it is overloaded?
What is the best way to solve this issue?
Thanks in advance.
Jijo
Hi All,
While doing 5 REGISTER/second, i observed that that avp_db_query is
returning data from a different db entry. This happens only with TCP with
more than 1 tcp_children. it works fine with UDP. It also work with TCP
with tcp_children as 1. Is there any relation between the TCP connections
and Postgres in kamailio. I'm using kamilio 3.1.0.
How should i debug this issue?? What are the areas need to look?
Thanks
Jijo
Hello,
during last days I spent some time to extend the native API exported to
Lua. Many more functions exported by core and modules can be called from
embedded Lua scripts. Note that you get also access to psedo-variable
operations and you can call all functions exported by modules that have
no fixup pretty safe, via sr.modf(...), without a need to be exported
natively.
I made an easy-to-do tutorial where all the SIP routing logic is
implemented in a Lua script (including authentication, accounting,
registrar, user location). You can read it at:
http://asipto.com/u/h
Therefore, if the configuration file language is not offering everything
you need to route your SIP traffic, look at app_lua module and its API,
it is a real option right now. Lua is a small and fast embeddable
language, the interpreter is linked to and loaded by Kamailio at statup,
therefore the performance penalty is not significant (Lua project site
is: http://www.lua.org). It is also popular among other SIP telephony
projects.
Any kind of feedback is appreciated!
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Trainings
Nov 22-25, 2010, Berlin, Germany
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com
LinkedIn
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Hi,
I have some troubles to send via t_uac_dlg a REGISTER method with an authorization field cause the values like digest, realm...are containing between double-quotes and that produce a parse error in kamailio.
I didn't find how can i escape those double-quotes to prevent from this error.
Here's the error logged:
Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: we have a quoted value, "From: <sip:131@192.168.40.150>To: <sip:131@192.168.40.150>Contact: <sip:131@192.168.40.210:53000>Authorization: Digest username="131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: " found p is "131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: we have reached the end of attr value, p is "131", realm="Asterisk/CM-YAHSAT", algorithm=MD5, uri="sip:192.168.40.202", nonce="5c87d7ed", response="fc126aadd3ba6b6ffbf38aea6be02122"Call-ID: b1ab82011346ff36(a)192.168.40.101CSeq: 10036 REGISTERExpires: 4294967295"Nov 22 15:11:07 [14258] DBG:mi_datagram:mi_datagram_parse_node: attr value <From: <sip:131@192.168.40.150>To: <sip:131@192.168.40.150>Contact: <sip:131@192.168.40.210:53000>Authorization: Digest username=> foundNov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_node: didn't find newline case1 Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_node: parse error around 1Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_parse_tree: parse error!Nov 22 15:11:07 [14258] ERROR:mi_datagram:mi_datagram_server: failed to parse the MI tree
regards,Koon
Daniel,
I can log into serweb.iptel.org with my username 'netlink83' and my mobile
number is saved in my account. I hope that is enough to have my mobile phone
registered. Maybe I should register to iptel again. What numbers can I call via
iptel? How do I configure my own pstn gateway? Is there instructions on
iptel.org?
Thanks,
Oscar Ying
650 776 9821
oscar.ying(a)yahoo.com
________________________________
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
To: oscar ying <oscar.ying(a)yahoo.com>
Cc: kamailio <sr-users(a)lists.sip-router.org>
Sent: Thu, November 18, 2010 2:02:18 AM
Subject: Re: [SR-Users] iphone
On 11/18/10 4:13 AM, oscar ying wrote:
>I've tried two different numbers and they both fail. I can make regular
>mobile calls but not VoIP calls via my SIP softphone (media5fone). What
>do you mean if my phone is registered?
>
A sip phone register to SIP server with a username and password - that is
what I mean with registered.
Note that iptel.org does not allow you to call to pstn, unless you have your
own pstn gateway configured. You can do voip calls for free.
Cheers,
Daniel
--
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Trainings
Nov 22-25, 2010, Berlin, Germany
Jan 24-26, 2011, Irvine, CA, USA
http://www.asipto.com
