Hi Daniel
Thanks for the reply! Here are things we use:
* keepalived for HA
* Ubuntu 12.04
There seems to be problems when Kamailio operates on both UDP and TCP at
the same time. But I'll upgrade Kamailio from v4.0.0. to v4.0.7 first and
see what we get, since there are a few bug fixes to do with TCP between
these two versions.
Cheers,
Yufei
On 4 April 2015 at 11:00, <sr-users-request(a)lists.sip-router.org> wrote:
> Send sr-users mailing list submissions to
> sr-users(a)lists.sip-router.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
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>
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of sr-users digest..."
>
>
> Today's Topics:
>
> 1. Re: SIP messages over UDP with sizes over MTU (Yufei Tao)
> 2. Re: SIP messages over UDP with sizes over MTU
> (Daniel-Constantin Mierla)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 3 Apr 2015 12:04:49 +0100
> From: Yufei Tao <yufei.tao(a)gmail.com>
> To: sr-users(a)lists.sip-router.org
> Subject: Re: [SR-Users] SIP messages over UDP with sizes over MTU
> Message-ID:
> <
> CAJwP0iTBGk8S91CheS+xAON2fJgyKZH9HFcwTbvcLygmzLNd+g(a)mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Daniel
>
> We don't have pike module enabled.
>
> When the problem occurred on an VIP, we observed these:
> * Kamailio stopped responding to any messages that were sent to the VIP,
> not just OPTIONS
> * The OPTIONS messages are not big. But the other SIP messages, e.g. some
> of the INVITE/OK that came from some SBCs can be big
> * netstat showed the Recv-Q on that VIP had a lot of bytes accumulated,
> while Kamailio was not seeing/reading them
> * Kamailio responded fine on its real IP, when I sent OPTIONS pings to it
> using sipsak
> * After restarting Kamailio it started working. But after 1 week or so the
> same problem happened again
>
> Since this only happened after running for 1 week or so, we didn't have any
> traces to show what exactly happened at the particular time when it
> happened. It is possible some SIP messages may have come in fragmented and
> some are just too big, depending on the route they came from etc. So I was
> wonder if it was possible that the UDP receive buffer was filled somehow
> with messed-up messages. Is there anyway to check this? Any suggestions
> where I should start looking please? Or is it generally a bad idea to use
> UDP when there are messages that may be too big, either fragmented or not?
>
> Since the it is running in the production environment, I'd like to get some
> confidence that a Kamailio upgrade will fix the problem first before I
> change anything there.
>
> Cheers,
> Yufei
>
>
>
> Date: Wed, 01 Apr 2015 16:27:44 +0200
> From: Daniel-Constantin Mierla <miconda(a)gmail.com>
> To: "Kamailio (SER) - Users Mailing List"
> <sr-users(a)lists.sip-router.org
> >
> Subject: Re: [SR-Users] SIP messages over UDP with sizes over MTU
> Message-ID: <551C0060.3010002(a)gmail.com>
> Content-Type: text/plain; charset="windows-1252"
>
> Hello,
>
> first, not related to the topic you reported, I would recommend at least
> upgrading to latest 4.0.x, there were many fixes from v4.0.0 and there
> are no changes to be done to config or database -- simply deploy latest
> 4.0.x and restart.
>
> Back to this topic. Is all the other traffic handled apart of big OPTIONS?
>
> Do you have pike module enabled? If yes, can you double check and be
> sure that the SBC is not blacklisted (traffic from it should not be
> handled via pike_check_req()).
>
> Cheers,
> Daniel
>
> On 01/04/15 15:42, Yufei Tao wrote:
> > Hi
> >
> > We've got Kamailio (v4.0.0) connected to some SBC, which sends SIP
> > traffic and periodic OPTIONS pings to Kamailio's VIP. Kamailio
> > responds to the OPTIONS pings with OK, i.e. in the main route block:
> > if (is_method("OPTIONS"))
> > {
> > sl_send_reply("200","OK");
> > exit;
> > }
> >
> > All works fine for a week or so, then Kamailio stopped responding to
> > the OPTIONS pings on the VIP it listens to. But it still respond to
> > OPTIONS pings that are sent to its real IP. The real IP is not used
> > for receiving/sending any traffic while only the VIP is. So it seems
> > that Kamailio is still working, but maybe having problems with the
> > receiving buffer for the VIP?
> >
> > We do see that some SIP messages sent to Kamailio's VIP are too big
> > (sometimes over 1500 bytes). My question is, in this case, what would
> > be expected to happen? Is it possible somehow the receiving buffer for
> > the VIP got messed up by the big UDP messages? Any one seen similar
> > problems? What is the suggested solution?
> >
> > We're considering moving to TCP. But since this is production
> > environment, I want to get some confidence that the problem we saw was
> > likely to have been caused by the UDP message being too large.
> >
> > Cheers,
> > Yufei
>
Thanks a lot Daniel. It was indeed the Cseq coming out of order from my box. Your comment helped me in identifying the problem.
Regards,
Badri.
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of sr-users-request(a)lists.sip-router.org
Sent: 03 April 2015 11:00
To: sr-users(a)lists.sip-router.org
Subject: sr-users Digest, Vol 119, Issue 3
Send sr-users mailing list submissions to
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You can reach the person managing the list at
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When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..."
Today's Topics:
1. Re: Make deb (Daniel-Constantin Mierla)
2. Re: [sr-dev] Announcement: Kamailio is now systemd-rtc-server
(Alex Balashov)
3. Re: Kamailio coredump when "INVITE" enter in failure route
(Jos? Seabra)
4. Re: Kamailio coredump when "INVITE" enter in failure route
(Daniel-Constantin Mierla)
5. Re: Upgrade from Sip Express Router 0.9.6 (Andres)
6. Dialog based CDRs and failed dialogs (Mickael Marrache)
7. Re: Upgrade from Sip Express Router 0.9.6
(Daniel-Constantin Mierla)
8. Re: Dialog based CDRs and failed dialogs
(Daniel-Constantin Mierla)
9. Re: Upgrade from Sip Express Router 0.9.6 (Andres)
10. Re: Dialog based CDRs and failed dialogs (Mickael Marrache)
11. Kamailio v4.2.4 Released (Daniel-Constantin Mierla)
12. Reg. invalid cseq error. (Badri Ranganathan)
13. Re: Reg. invalid cseq error. (Daniel-Constantin Mierla)
14. Re: Dialog based CDRs and failed dialogs
(Daniel-Constantin Mierla)
----------------------------------------------------------------------
Message: 1
Date: Thu, 02 Apr 2015 12:07:06 +0200
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
To: jay binks <jaybinks(a)gmail.com>
Cc: "Kamailio \(SER\) - Users Mailing List"
<sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] Make deb
Message-ID: <551D14CA.4030201(a)gmail.com>
Content-Type: text/plain; charset="utf-8"
Do you see db_cassandra being compiled in the logs printed by 'make deb'?
Cheers,
Daniel
On 02/04/15 03:37, jay binks wrote:
> I have managed to confirm that "make deb" does not actually build
> db_cassandra but it does make a deb for it :) ( has changelog etc but
> no db_cassandra.so file )
>
> why does it not build the .so when calling make deb ??
> can someone point me in the direction to debug that part.
>
> doesnt seem to be related to the debian rules or control file ..
>
> is there something like modules.lst for when using "make deb" ?
>
>
> On 2 April 2015 at 00:24, Daniel-Constantin Mierla <miconda(a)gmail.com
> <mailto:miconda@gmail.com>> wrote:
>
> The spec files to build debian packages are kept inside
> 'pkg/kamailio/deb/' directory, groupped per distribution name
> inside specific sub-directories (using debian and ubuntu version
> names -- the one 'debian' is the generic one, but for a specific
> version you should choose the one matching the name).
>
> Then I am not that familiar with debian spec files to be able to
> advice by heart, however you should find easily on the web many
> sources of information about the files inside those folder and how
> to build deb packages.
>
> Say you want to build for debian wheezy:
>
> - edit the files inside pkg/kamailio/deb/wheezy/ to fit your needs
> - in the root directory with kamailio sources do:
>
> ln -s pkg/kamailio/deb/wheezy debian
> make deb
>
> And you should get the packaging process started.
>
> Cheers,
> Daniel
>
>
> On 01/04/15 14:46, jay binks wrote:
>> Sorry, this confused me a little..
>> can you give an example of what you mean ?
>>
>> On 1 April 2015 at 22:40, Daniel-Constantin Mierla
>> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
>>
>> Hello,
>>
>> you have to edit spec files and remove the other modules in
>> the folder matching your debian distribution inside
>> 'pkg/kamailio/deb/'. This is the only way I know to change
>> the packaging content for debs. Then make a symlink inside
>> root folder of kamailio server named 'debian' to the
>> respective folder inside 'pkg/kamailio/deb/' and run 'make deb'.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 01/04/15 13:50, jay binks wrote:
>>> is there any way to make deb packages for only 1 of the
>>> modules ??
>>> currently it takes a long time to run "make deb"
>>>
>>> im simply trying to build and package db_cassandra ( need
>>> the package for my environment )
>>>
>>> As a side note, is there any chance db_cassandra could be
>>> added to the kamailio deb packages ?
>>>
>>> --
>>> Sincerely
>>>
>>> Jay
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>> --
>> Daniel-Constantin Mierla
>> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda
>> Kamailio World Conference, May 27-29, 2015
>> Berlin, Germany - http://www.kamailioworld.com
>>
>>
>> _______________________________________________
>> 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
>>
>>
>>
>>
>> --
>> Sincerely
>>
>> Jay
>
> --
> Daniel-Constantin Mierla
> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda
> Kamailio World Conference, May 27-29, 2015
> Berlin, Germany - http://www.kamailioworld.com
>
>
>
>
> --
> Sincerely
>
> Jay
--
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
I've been working on Kamailio websocket integration and I believe I'm having
issues with the IPv6 address representation in the Contact header's alias
parameter. After Googling, it appears after
https://github.com/kamailio/kamailio/commit/814c08f3 the IPv6 contact is
represented in brackets []. However, it when using websockets, and the alias=
parameter uses brackets [] around and IPv6 address, there are message parsing
issues.
With a header such as the following
Contact:
<sip:wstest1@example.com;gr=urn:uuid:26140e27-0ab7-4e65-98e3-3d0909b1434e;alias=[2001:db8:0:1]~48768~6>
Asterisk 13.2.0 will give the following error:
pjsip:0 <?>: sip_transport. Error processing 1855 bytes packet from UDP
10.77.79.3:5060 : PJSIP syntax error exception when parsing 'Request Line'
header on line 12 col 129:
And when sipjs, or jssip are used with either Firefox or Chrome, they send
garbage in the ACK request URI:
Kamailio logs something like the following and the ACK cannot be processed:
WARNING: sanity [sanity.c:236]: check_ruri_scheme(): failed to parse request
uri [�a�{1me▒s�na@50�9���1.�8:�2v0;i��a��=11>7�n7x>10O�2v0~1]
Is it proper to have [] brackets around the IPv6 alias address in the Contact
header? Does the value need to be quoted?
If I force the browser to use IPv4, without changing anything else, both jssip
and sipjs work perfectly in Firefox and Chrome.
--
Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
Pull requests are welcome, guys. We are hitting 30k rtp sessions on one of
our largest clusters. Lot of changes are coming into git repo soon to make
rtp_cluster component run smoothly under such conditions.
On Mar 16, 2015 8:56 PM, "John Mathew" <john.mathew(a)divoxmedia.com> wrote:
> Yes
>
> On Tuesday, 17 March 2015, Zheng Frank <zhengyumingapple(a)gmail.com> wrote:
>
>> Do you mean ROHC ?
>>
>> 2015-03-14 12:39 GMT+08:00 Maxim Sobolev <sobomax(a)sippysoft.com>:
>>
>>> Do you have any particular RFC in mind?
>>> On Mar 12, 2015 10:28 AM, "John Mathew" <john.mathew(a)divoxmedia.com>
>>> wrote:
>>>
>>>> Hi,
>>>>
>>>> Maxim,
>>>> Is there any plans for rtp header compression in future. I can't see
>>>> anything in the change log for 2.0.0
>>>>
>>>>
>>>> On Tuesday, 10 March 2015, Maxim Sobolev <sobomax(a)sippysoft.com> wrote:
>>>>
>>>>> Hi All,
>>>>>
>>>>> I'm happy to announce that we have released rtpproxy v2.0.0.
>>>>>
>>>>> You can review the release notes here:
>>>>> https://github.com/sippy/rtpproxy/releases/tag/v2.0.0
>>>>>
>>>>> -sobomax
>>>>>
>>>>>
>>>>
>>>> --
>>>> Sent from iPhone 6
>>>>
>>>> --
>>>> You received this message because you are subscribed to the Google
>>>> Groups "rtpproxy" group.
>>>> To unsubscribe from this group and stop receiving emails from it, send
>>>> an email to rtpproxy+unsubscribe(a)googlegroups.com.
>>>> To post to this group, send email to rtpproxy(a)googlegroups.com.
>>>> To view this discussion on the web visit
>>>> https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B…
>>>> <https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpUhu9fpwmqpRFiaXsQo9_%2B…>
>>>> .
>>>> For more options, visit https://groups.google.com/d/optout.
>>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>
> --
> Sent from iPhone 6
>
> --
> You received this message because you are subscribed to the Google Groups
> "rtpproxy" group.
> To unsubscribe from this group and stop receiving emails from it, send an
> email to rtpproxy+unsubscribe(a)googlegroups.com.
> To post to this group, send email to rtpproxy(a)googlegroups.com.
> To view this discussion on the web visit
> https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpWvmaQnE8qM8ibb10d%3DNj9…
> <https://groups.google.com/d/msgid/rtpproxy/CA%2BSkwpWvmaQnE8qM8ibb10d%3DNj9…>
> .
> For more options, visit https://groups.google.com/d/optout.
>
Hi all,
I make calls towards the Kamailio from my call generator and I get a message saying "invalid cseq error" sometimes at the 1005th registration or sometimes at the 1013thregistration, but the cseq number seems to be fine. Any idea about why this would happen?
Thanks,
Badri.
How would one even approach the daunting task of upgrading a fully
operational Sip Express Router 0.9.6 installation with thousands of
users that has been running unmodified for over 10 years.
I would like some pointers on how to approach the mysql database
migration from the old schema to the Kamailio 4.2 schema. I sure hope I
don't have to start from scratch. Maybe somebody has already done an
upgrade script?
Regarding the config file, I will certainly have to rewrite it but I was
wondering if there was a list of deprecated commands or syntaxes that
would no longer be valid and their equivalent replacement.
Thanks,
--
Technical Support
http://www.cellroute.net
Hello all,
I have RCS based client.
I would like to implement MUC(multi chat user) in existing Kamailio IMS
Server.
Please help me to input the procedure.Is it really possible to do it ?
Please share me URL for any existing "C" source code which I can directly
imbibe into Kamailio IMS serve
--
View this message in context: http://sip-router.1086192.n5.nabble.com/Regarding-multi-user-chat-in-Kamail…
Sent from the Users mailing list archive at Nabble.com.
Hi.
I think there is a bug in function search_hf_f() in textops.c
Here is what I do
if ( method == SUBSCRIBE ) {
if ( !search_hf("Event", "call-completion", "f") ) {
sl_send_reply("489","Bad Event");
exit;
}
}
But here is a SUBSCRIBE that do pass the proxy.
SUBSCRIBE sip:101@192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP
77.10.66.72:59793;branch=z9hG4bKPj.O5IXVtDsImPNfCGnDkvj91INsEfvZsm
Max-Forwards: 70
From: <sip:101@192.168.1.50>;tag=8-cXkrmwx5WGSIAYgYaXW0oTTsEAEM1S
To: <sip:101@192.168.1.50>
Contact: <sip:101@192.168.1.8:59793;ob>
Call-ID: O12dPIx7Wiih2b.o36UFmMdXr6RE-8Jl
CSeq: 24359 SUBSCRIBE
Route: <sip:192.168.1.50;transport=udp;lr>
Event: message-summary
Expires: 3600
Supported: replaces, 100rel, timer, norefersub
Accept: application/simple-message-summary
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple_mako-21/r2457
Content-Length: 0
As you can see the Event header doesn't contain "call-completition".
Looking at function search_hf_f() i saw that if nothing is found value
-1 will be returned. if there is the match then 1 is returned. (see line
2779 and 2797)
But in line 2782 the function will return 1 instead of -1
index 50a6f07..a945937 100644
--- a/modules/textops/textops.c
+++ b/modules/textops/textops.c
@@ -2779,7 +2779,7 @@ static int search_hf_f(struct sip_msg* msg, char*
str_hf, char* re, char *flags)
return 1;
} else {
if(flags!=NULL && *flags=='f')
- return 1;
+ return -1;
}
} else {
hfl = hf;
I am no programmer so it's possible that I misunderstood something.
greetz
Paolo
