Hi guys,
i have yet to finish my readings on the websocket standards but just wanted
to fire away with this question.
is the behavior of protocol conversions between UDP and TCP the same as if
you include websockets?
i have this twinkle issue (an old SIP stack)
Received from: udp:192.168.122.100:5060
INVITE sip:kelvin@192.168.122.1 SIP/2.0
Record-Route: <sip:192.168.122.100;r2=on;lr=on>
Record-Route: <sip:192.168.122.100:8080;transport=ws;r2=on;lr=on>
Via: SIP/2.0/UDP
192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0
Via: SIP/2.0/WS
z173czhz21tk.invalid;rport=54765;received=192.168.122.1;branch=z9hG4bK3818745
Max-Forwards: 16
To: sip:kelvin@192.168.122.100
From: sip:kelvin2@192.168.122.100;tag=lmf8ofkxwq
Call-ID: 69hbgnng64at9p07r2j4
CSeq: 9406 INVITE
Contact: <sip:kelvin2@z173czhz21tk.invalid
;alias=192.168.122.1~54765~5;transport=ws;ob>
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, MESSAGE, SUBSCRIBE
Content-Type: application/sdp
Supported: path, outbound, gruu
User-Agent: JsSIP 0.2.1
Content-Length: 2103
v=0
o=- 3148117784 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
m=audio 55736 RTP/SAVPF 103 104 111 0 8 106 105 13 126
c=IN IP4 192.168.122.1
a=rtcp:55736 IN IP4 192.168.122.1
a=candidate:2625852906 1 udp 2113937151
<cut off>
---
+++ 16-3-2013 10:41:25.584732 INFO SIP ::send_sip_udp
Send to: udp:192.168.122.100:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS
z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745
To: <sip:kelvin@192.168.122.100>
From: <sip:kelvin2@192.168.122.100>;tag=lmf8ofkxwq
Call-ID: 69hbgnng64at9p07r2j4
CSeq: 9406 INVITE
Server: Twinkle/1.4.2
Content-Length: 0
---
+++ 16-3-2013 10:41:25.589231 INFO SIP ::send_sip_udp
Send to: udp:192.168.122.100:5060
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP
192.168.122.100;branch=z9hG4bKed9e.8b1b2fa61a90a9031e17b393657df31b.0,SIP/2.0/WS
z173czhz21tk.invalid;received=192.168.122.1;rport=54765;branch=z9hG4bK3818745
To: <sip:kelvin@192.168.122.100>;tag=pxtmo
From: <sip:kelvin2@192.168.122.100>;tag=lmf8ofkxwq
Call-ID: 69hbgnng64at9p07r2j4
CSeq: 9406 INVITE
Server: Twinkle/1.4.2
Warning: 302 X340precise "Incompatible transport protocol"
Content-Length: 0
Kelvin Chua
Good day.
How could I debug SIP dialogs?
--
SY,
Victor
JID: coyote(a)bks.tv
JID: coyote(a)bryansktel.ru
I use FREE operation system: 3.8.2-calculate GNU/Linux
Hello,
Thank you for your reply
I follow your suggestions to modify SHM_MEMORY ,PKG_MEMORY value and
the system only two uac register.
See the program after I start kamailio.
shm memory 1G
pkg memory 10M
sh-3.00# ps -ef | grep kamailio
root 23627 1 0 12:57 ? 00:00:00
/home/pkg/kamailio-3.3.4/sbin/kamailio -f
/home/pkg/kamailio-3.3.4/etc/kamailio/kamailio.cfg -P
/var/run/kamailio/kamailio.pid -m 1024 -M 10 -u root -g kamailio
root 23635 23627 0 12:57 ? 00:00:03
/home/pkg/kamailio-3.3.4/sbin/kamailio -f
/home/pkg/kamailio-3.3.4/etc/kamailio/kamailio.cfg -P
/var/run/kamailio/kamailio.pid -m 1024 -M 10 -u root -g kamailio
This problem is still the same last crazy the sending options message for uac
But the /var/log/syslog not the same as follows
Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]:
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
tl=0x77052ae4 tl->next=(nil) tl->prev=(nil)
Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]:
ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime
Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]:
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
tl=0x77052ae4 tl->next=(nil) tl->prev=(nil)
Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]:
ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime
Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]:
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
tl=0x77052ae4 tl->next=(nil) tl->prev=(nil)
Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]:
ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime
Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]:
CRITICAL: dialog [dlg_timer.c:205]: Trying to update a bogus dlg
tl=0x77052ae4 tl->next=(nil) tl->prev=(nil)
Mar 20 12:58:53 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[23637]:
ERROR: dialog [dlg_req_within.c:231]: failed to update dialog lifetime
2013/3/20 Rinor Hoxha <rinorhoxha(a)gmail.com>
> Hi,
> I had this "issue" some times ago. It turned out that I needed to increase
> the value of SHM_MEMORY, since by mistake I forgot a 0 on the value (1024
> vs. 10240). However this happened only when high loads on system.
> I needed also to tune PKG_MEMORY to my needs. ( both on
> /etc/default/kamailio )
>
> Check your settings and see if this helps.
>
> Br, Rinor
>
>
>
> On Tue, Mar 19, 2013 at 4:07 PM, dolphin <dolphinctk(a)gmail.com> wrote:
>
>> hi ,
>>
>> i use kamailio 3.3.4 version
>>
>> I set and load dialog module
>>
>> modparam("dialog", "ka_timer", 10)
>> modparam("dialog", "ka_interval", 20)
>>
>> use dlg_set_property("ka-src");
>>
>> When uac A call uac B connection success , After 20 seconds, sip proxy
>> send the OPTIONS message uac A
>>
>> Crazy sent to fill my bandwidth , I use ngrep tools see sip server
>>
>> U xxx.xxx.xxx:5002 -> xxx.xxx.xxx.xxx:16296
>> OPTIONS sip:xxxxxx@xxx.xxx.xxx.xxx:16296;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5002;branch=z9hG4bK92da.1584702.0.
>> To: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=eb6e8f3e.
>> From: sip:xxxxxxx@xxx.xxx.xxx.xxx
>> :5002;tag=D.ZHkBjEnKXX.DTO69bc09sbB4oFdU3O.
>> CSeq: 1 OPTIONS.
>> Call-ID: NDZlY2JhMjUyMjZiYTllZGJmZTA2MmYxODU5NmY3NGY..
>> Content-Length: 0.
>> User-Agent: SIP Server.
>>
>> and /var/log/syslog
>>
>> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
>> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370)
>> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
>> tm [uac.c:338]: t_uac: Error while building message
>> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
>> dialog [dlg_req_within.c:381]: failed to send the BYE request
>> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
>> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370)
>> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
>> tm [uac.c:338]: t_uac: Error while building message
>> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
>> dialog [dlg_req_within.c:381]: failed to send the BYE request
>> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
>> tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370)
>> Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
>> tm [uac.c:338]: t_uac: Error while building message
>>
>> Please help me
>>
>>
>>
>>
>> _______________________________________________
>> 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
>>
>
Thanks for the idea.
After sending the original email I got user2 to call through to user1. Zrtp even was negotiated. So I think usrloc is sort of sharing the registration information but sometimes the call doesn't get sent through under certain circumstances.
Maybe it's because user1 is behind a nat and rtpproxy is being used to keep its connection alive while user2 is not nat'd.
I think it is almost working.
-----Original Message-----
From: David | StyleFlare <david(a)styleflare.com>
Date: Wed, 20 Mar 2013 03:27:06
To: <sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] 2 kamailio servers sharing one database and one dns
name
I dont think it works like that, obviously the server 1 does not have a connection if the user is registered on server 2.
On 3/19/13 11:05 PM, David Thomson wrote:
hi,
I have two kamailio 3.3.4 servers sharing one database. usrloc module is loaded on both machines. The DNS name for the machines is shared (i.e siptest.testdomain.com with 2 public ip's - one for Server1 and one for Server2) and setup in round robin mode.
The scenario is as follows:
User 1 registers to Server1
User 2 registers to Server2
User1 tries to call User2 but Server1 throws an error and the call doesn't ever connect to User2:
WARNING: usrloc [udomain.c:321]: non-local socket <udp:xxx.xxx.xxx.xxx:5060> <udp:xxx.xxx.xxx.xxx:5060> ...ignoring.
Any ideas?
ttyl,
Dave
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
I don't know if it's something wanted, but the email header "List-Id"
has changed for both sr-dev and sr-users.
The modifications :
List-Id: "SIP Router - Kamailio" -> List-Id: "Kamailio \(SER\) - Users
Mailing List"
List-Id: "Development mailing list of the sip-router project" ->
List-Id: "Kamailio \(SER\) - Development Mailing List"
Regards,
--
Nikita
hi ,
i use kamailio 3.3.4 version
I set and load dialog module
modparam("dialog", "ka_timer", 10)
modparam("dialog", "ka_interval", 20)
use dlg_set_property("ka-src");
When uac A call uac B connection success , After 20 seconds, sip proxy
send the OPTIONS message uac A
Crazy sent to fill my bandwidth , I use ngrep tools see sip server
U xxx.xxx.xxx:5002 -> xxx.xxx.xxx.xxx:16296
OPTIONS sip:xxxxxx@xxx.xxx.xxx.xxx:16296;transport=udp SIP/2.0.
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5002;branch=z9hG4bK92da.1584702.0.
To: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=eb6e8f3e.
From: sip:xxxxxxx@xxx.xxx.xxx.xxx:5002;tag=D.ZHkBjEnKXX.DTO69bc09sbB4oFdU3O.
CSeq: 1 OPTIONS.
Call-ID: NDZlY2JhMjUyMjZiYTllZGJmZTA2MmYxODU5NmY3NGY..
Content-Length: 0.
User-Agent: SIP Server.
and /var/log/syslog
Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370)
Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
tm [uac.c:338]: t_uac: Error while building message
Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
dialog [dlg_req_within.c:381]: failed to send the BYE request
Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370)
Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
tm [uac.c:338]: t_uac: Error while building message
Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
dialog [dlg_req_within.c:381]: failed to send the BYE request
Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
tm [t_msgbuilder.c:1528]: build_uac_req(): no shmem (370)
Mar 19 21:09:07 cfa /home/pkg/kamailio-3.3.4/sbin/kamailio[12237]: ERROR:
tm [uac.c:338]: t_uac: Error while building message
Please help me
Hi all,
I’m working for a while with Kamailio+Freeswitch as SBC.
I have this structure:
* *
* *
* *
* *
* *
* *
* *
* *
* *
When I make a call from one client connected to LCR, it is route to my SBC
and afterwards to his destiny in the cloud passing though a gateway. When
the destiny is unreachable, the LCR reroutes the call to another gateway.
Sometimes the LCR send this retry again to the SBC , because the second
gateway is also in the cloud, but when this happens, FreeSWITCH answer with
:
“482 Request merged” because it detects that is the same call.
This is because the second INVITE has the same Call-ID and same Cseq.
Kamailio *is not* increasing CSeq.
Is there a way to resolve this?
Regards,
*Camila Troncoso **|* Ingeniero de Desarrollo
RedVoiss *|*ctroncoso(a)redvoiss.net
Santiago - Chile *|* +56 2 2408535
www.redvoiss.net
Daniel,
Thanks for the help. YES after disabling TLS for Kamailio, it is working
fine.
Is there a solution to have both TLS for Kamailio and SSL for LUA?
Thanks
Krish Kura
Forgive me if I failed to follow a recent discussion to its conclusion,
but what is the bottom line on calling uac_replace_from() consecutively,
in the course of processing one request?
Is it possible, while using restore_mode 'auto'? Is there a correct way
to do it, such as calling msg_apply_changes() after the first call but
not the second one?
If not, what is the alternative?
-- Alex
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
