Hi Austin,
Kamailio should be pretty fast on these things.
Here are things which you can check:
* Increase number of TCP workers/make sure you are using async TCP.
* Use in-memory store for your registrations, not DB ones (at least
dump async your contacts in DB).
* Make sure you allow enough files to be opened in case of TCP
connections (I suspect this is your issue there).
Regarding performance tests, not sure if you saw already these links:
http://www.kamailio.org/docs/openser-performance-tests/https://sip-router.org/wiki/performance/v3.0-capacity
Hope these help!
DanB
Hello,
I noticed that Kamailio adds a Server header in the replies generated by it, is there anyway to disable it?
Thanks in advance for your help,
Nitesh
Greetings.
I want to modify INVITE answers processing in this way:
1) receive INVITE answer
2) check if recipient is registered but has no TCP/TLS connection now
3) suspend answer for 30 seconds (for example)
4) resume answer if REGISTER comes from lost client in these 30 seconds
5) terminate INVITE session if 30 seconds are expired
What I do not understand - is it possible to find if client is
connected now to kamailio?
I talk about TCP/TLS connection only - not about registration record.
Thank you!
Hello,
I am trying to play with domain names in the Record-Route header.
This is for a case when kamailio is doing the translation between UDP and TLS, so
it is doing double record routing.
I want to use the ip addresses on inbound UDP leg and
domain names on the outbound TLS leg in record-route headers.
I'm doing this by calling the following function
-- record_route_preset("xyz.com;transport=tls", "1.1.1.1"); ----
This does the trick of adding a Record-Route in initial TLS invite on the outbound leg:
-----------------
Record-Route: <sip:xyz.com;transport=tls;lr=on>
Record-Route: <sip:1.1.1.1;lr=on>
----------------
My problems start when the calling party sends a ACK to kamailio over UDP
with the Route header
Route: <sip:1.1.1.1;lr=on>,<sip:xyz.com;transport=tls;lr=on>
I call loose_route() function in config file and as per my expectation, Kamailio should see that both
these Route headers belong to Kamailio and it should use Request URI for forwarding the ACK,
but unfortunately, Kamailio thinks that xyz.com is some other entity and tries to forward the ACK
to xyz.com doing a DNS query.
What can i do to avoid this problem, i need kamailio to recognize that "xyz.com" is kamailio as i
have a defined an alias in the config file too as follows:
alias=xyz.com:5060
Thanks for your help in advance,
Nitesh
Hi!
I just found out that there is a new rtpproxy release:
http://www.rtpproxy.org/post/v2release/
Has anybody tested it and want to share some experiences? Or have people
turned to rtpengine meanwhile? (I am still using rtpproxy 1.2 as I do
not need new features).
regards
Klaus
Hi Daniel
I have rtpproxy to perform media relaying.
The case where it just timeouts:
1. Client A behind NAT (wifi) calling Client B on 3G
Sent from Samsung Mobile
-------- Original message --------
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date:05/05/2015 12:52 (GMT+05:30)
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] Kamailio proxy for far-end nat traversal
Hello,
do you have rtpproxy or rtpengine for relaying rtp packets? Kamailio is routing only sip packets, you need the rtp relay application to help with media streams.
Cheers,
Daniel
On 01/05/15 13:15, rahul.ultimate wrote:
Hello
I need a small guidance on creating a light weight proxy which only forwards the msgs to my sip server and also does supports symmetrical nated clients.
The way I have created the configuration is a slight modification of :
https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg
Problem is unless the sip clients use STUN media packets are not routed. And sometimes even the signalling does not pass through.
Should i not use fix_nated_register and switch to fix _ nated_contact always ?
Because in all those cases where signalling does not pass through, I see RTO. My best guess is kamailio trying to communicate with private ip.
Anything which I can try that you could suggest from the top of yourmind woyld be muc appreciated.
Thanks
Rahul
Sent from Samsung Mobile
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
Hi all, I just experienced a sudden crash from one of my Kamailio servers. I'm using v4.2.4 built from git repository (branch remotes/origin/4.2). The error logs show this at the time of the crash (I'm using debug=1 on config file)
May 15 15:59:53 KAMAILIO-ERS1 kamailio[21162]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 19
May 15 15:59:53 KAMAILIO-ERS1 kamailio[21141]: ALERT: <core> [main.c:784]: handle_sigs(): child process 21149 exited by a signal 11
May 15 15:59:53 KAMAILIO-ERS1 kamailio[21141]: ALERT: <core> [main.c:787]: handle_sigs(): core was not generated
May 15 15:59:53 KAMAILIO-ERS1 kernel: [2156637.186585] kamailio[21149]: segfault at 20b6 ip 00000000000020b6 sp 00007ffeb0c2bc88 error 14 in kamailio[400000+3c3000]
May 15 15:59:53 KAMAILIO-ERS1 kernel: [2156637.214933] kamailio[21141]: segfault at 2290 ip 0000000000002290 sp 00007ffeb0c2c4d8 error 14 in kamailio[400000+3c3000]
The modules I'm using are:
loadmodule "mi_fifo.so"
loadmodule "db_mysql.so"
loadmodule "sqlops.so"
loadmodule "tm.so"
loadmodule "sl.so"
loadmodule "pv.so"
loadmodule "xlog.so"
loadmodule "maxfwd.so"
loadmodule "siputils.so"
loadmodule "sanity.so"
loadmodule "textops.so"
loadmodule "rr.so"
loadmodule "statsd.so"
Have you noticed this error before? This server has a very simple config, for every invite received, it queries a MySQL database to replace the called number from 3 digits to 8-10 digits (for example 135 is translated to 1152751135) and sends back a 300 Redirect with the new number.
Another thing to notice is that this server was working at 3-4% CPU load and when the issue happen it spiked to 53%...
Let me know I can provide any other info that could be useful.
Thanks!
Federico
Federico San Martín
Ingeniería de Telefonía
TeleCentro S.A.
Cnel. Apolinario Figueroa 254 Piso 4to
CABA - C1414EDF
Fijo : +54 11 3977 10 45
Móvil : +54 9 11 5824 3240
e-mail : fsanmartin(a)telecentro.net.ar
Hello,
I am forwarding an incoming call to a destination and on time-out forward
to another and them another.
I need to send a '181 Call Is Being Forwarded' to my SBC to reset the early
media stream when forwarding to the next destiation which I can do
with sl_send_reply("181","Call
Is Being Forwarded").
However the SBC vendor requires that the To-tag in the 181 is cleared for
this to work.
How can I clear the To-tag from this 181 response?
Thanks
John
I need some instructions on how to setup dbtext properly
I generated dbtext files through "./kamdbctl create"
I then add user through "./kamctrl add user1 user1"
It seems to have added the user because when I do the show './kamctl show user1' command it displays
[1, ' user1', '10.1.1.1', ' user1', '', 'ee2820706921ce44a3f760e03ff19a22', '9e2c9e7df27f474d2b995c7e94ab32f2', '']
However viewing the subscriber file /usr/local/etc/kamailio/dbtext/subscriber
id(int,auto) username(string) domain(string) password(string) email_address(string,null) ha1(string,null) ha1b(string,null) rpid(string,null)
There is no entry for 'user1'.
Where is it storing the user data and why is it not in the subscriber file?.
user1 is not recognised when trying to register
3(20432) DEBUG: db_text [dbt_api.c:80]: dbt_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7f9e0dd776c0 3(20432) DEBUG: auth_db [authorize.c:196]: get_ha1(): no result for user 'user1@'
Regards
