HI Daniel,
Can you please share , why kamailio couldn't able to extract
the body, when INVITE comes with multi party body (sdp + resource-lists). Is
there any other way to extract the same.
Please share your views.
Regards
Surendra
Some udp INVITEs hit our network very close to the 1500 byte limit, and
we add 2 Record-Routes before we pass the INVITE to the end user. The
packet gets truncated in the SDP; "usually" it still works.
In our use case, I can't use gzip or TCP. We are already using textops
to convert to short-form headers where possible; we're also stripping
unnecessary headers.
So the next thing is to use RFC 3261 7.3 and 25.1 to concatenate the
multiple Record-Routes into one longer line (same for Via, Contact, etc.):
Record-Route: <sip:111.111.111.111;lr=on>
Record-Route: <sip:99.99.99.99;lr=on>
Record-Route: <sip:88.88.88.88:5060;lr>
Record-Route: <sip:77.77.77.77;lr>
becomes:
Record-Route: <sip:111.111.111.111;lr=on>,
<sip:99.99.99.99;lr=on>,
<sip:88.88.88.88:5060;lr>
<sip:77.77.77.77;lr>
That saves about 30 bytes - one can save more if putting all the values
on 1 line without the cr\lf between.
Doesn't look like there's any kamailio function in textops/x that makes
this kind of thing easy; so before I start hacking away at the code...
Our downstream uses freeswitch, and is willing to verify it works for
him - but from a larger interop view, is this a Bad Idea(tm)? Anyone
have any experience with brand-name carrier SBC/Proxies that would choke
on a header like that?
Thanks!
Trying to call a T.38 enabled endpoint B from an endpoint A that doesn't. This
results in a "488 Not Acceptable" from A.
The kamailio (4.0.3) in between with topoh enabled (with mask_callid set to 1.
Kamailio ACKs the 488 to A, but the ACK has the wrong (masked) Call-ID,
resulting in the ACK to A being ignored, A keeps sending 488 at intervals
before just dropping the call.
My understanding is the Call-ID in the ACK has to be the same as the Call-ID
from the 488. But instead of sending the unmasked Call-ID used in all other
messages between A and Kamailio, the ACK contains the masked Call-ID used in
all messagea between A and Kamailio.
I can't find a bug report relating to this. Is this a known feature or maybe
fixed in newer versions?
A->Kamailio
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/UDP
109.235.32.40;branch=z9hG4bKf91a.c6c88747.0;received=109.235.32.40.
Via: SIP/2.0/UDP 172.19.162.1;branch=pocos-
rS4MusXox1l5QHyNxRy6uAXsEOdsxidSEAktxGZKWgeKCgeS-
RrKEAy057Nl9DEpWpefZD6AhGtwWDJjEqeuEAZgZs5iZA7*.
From: <sip:+31880100799@sip.pocos.nl:5060>;tag=as3869fe2a.
To: "+31880100705" <sip:+31880100705@sip.pocos.nl>;tag=as1b0b8097.
Call-ID: 0b5946b977210450571f767a19cd6fa0(a)99sip.pocos.nl.
CSeq: 102 INVITE.
Kamailio<-A
ACK sip:+31880100705@109.235.32.48 SIP/2.0.
Via: SIP/2.0/UDP 109.235.32.40;branch=z9hG4bKf91a.c6c88747.0.
Max-Forwards: 16.
From: <sip:+31880100799@sip.pocos.nl:5060>;tag=as3869fe2a.
To: "+31880100705" <sip:+31880100705@sip.pocos.nl>;tag=as1b0b8097.
Call-ID: !!:xqXtWRMpZAngEsX3xGMtxGrgxDZgEA9JxR4AzGz8ZRIyWR4sh.yomqlACgxoC8K*.
CSeq: 102 ACK.
--
Telefoon: 088 0100 700
Sales: sales(a)pocos.nl | Service: servicedesk(a)pocos.nl
http://www.pocos.nl/ | Croy 9c, 5653 LC Eindhoven | Kamer van Koophandel
17097024
Hello,
given the roots and the tight relation of Kamailio project with the
academic environment, we are offering three seats at Kamailio World
Conference, May 27029, 2015, in Berlin, to students enrolled in
universities or research institutes (both bachelor and PhD programs
qualify).
Last year we tried it locally with the universities in Berlin and this
year we want to extend it, as there might be young people interested to
travel a bit and attend the event.
If you are a student and want to participate, email me directly
(miconda(a)gmail.com) or to registration(a)kamailio.org . Participation to
all the content of the event (workshops, conference and social event) is
free, but you will have to take care of expenses for traveling and
accommodation. Write a short description about your interest in real
time communications and what is the university or the research institute
you are affiliate to.
Also, if you are not a student, but you are in touch with some or have
access to students forums/mailing lists, it will be very appreciated if
you forward these details.
More information about Kamailio World is available on the web site:
- http://www.kamailioworld.com
Cheers,
Daniel
--
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 Sharath
Thanks for the tip! In fact just last night I've found what got things
stuck was nothing to do with Kamailio - it was one of our own little module
(plugged into Kamailio) that's got locked up.
The first thing I suspected was like what you mentioned that Kamailio got
stuck due to fragmentation. I wonder if you've seen this happening with
Kamailio? At least in this problem of ours it wasn't Kamailio.
Thank you and Daniel for the help on this!
Yufei
On 22 April 2015 at 22:19, Sharath Kumar <Sharath.Kumar(a)mezocliq.com> wrote:
> It is definitely possible due to fragmentation the sip receiver was
> stuck in some weird state. If I were you I would do TCP for SIP. It is also
> recommended in 3261 to use TCP when sip is that big.
> ------------------------------
> *From:* sr-users <sr-users-bounces(a)lists.sip-router.org> on behalf of
> Yufei Tao <yufei.tao(a)gmail.com>
> *Sent:* Friday, April 3, 2015 7:04 AM
> *To:* sr-users(a)lists.sip-router.org
>
> *Subject:* Re: [SR-Users] SIP messages over UDP with sizes over MTU
>
> 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
>
Hello,
development (master) branch of Kamailio enters now in pre-release phase
for version 4.3.0. Therefore, no new feature should be pushed to master
until we create a dedicated branch for 4.3 (expected to be in about 4
weeks or so).
If in doubt to push or not a commit to master, push it first on a
personal branch (or attach to an email) and discuss it on sr-dev. The
new modules can be a bit more dynamic if there is need to get them to
the right shape (e.g., like decision to rename functions, parameters or
adjust database structure).
We hope to get many people involved in testing, to rich a stable state
before releasing 4.3.0. If you want to get involved and need assistance,
don't hesitate to write to mailing lists.
Help with updating the wiki page for migration from 4.2 to 4.3 as well
as what is new in 4.3 is very appreciated. I will post updates about the
very soon.
Many thanks to everyone involved in development of 4.3 and the early
testers that played with master branch so far.
Cheers,
Daniel
--
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
Hello everyone,
I’m currently running Kamailio 4.2.4 with PostrgreSQL 9.3 database, each one on CentOS 6.6
I have installed and configured Kamailio with PostgreSQL without any problem. But now, I don’t want my phones register be recorded on Kamailio (on cache) but immediately on the external database.
For that, I follow your advices and configure the kamailio.cfg this way :
I added these lines :
#!ifdef WITH_PGSQL
#!define DBURL "postgres://user:passwd@database.domain:5432/kamailio"
#!endif
…
#!ifdef WITH_PGSQL
loadmodule "db_postgres.so"
#!endif
…
And I have all these modules laoded :
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "corex.so"
loadmodule "tm.so"
loadmodule "tmx.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "outbound.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so"
loadmodule "acc.so"
loadmodule "stun.so"
I also set 3 to the usrloc param so my registration are directly on my database :
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 3)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
Then, when I restart my Kamailio server, I saw on my postgres database that the connection with this user and passwd is ok but my sip phones didn’t registered. Thanks to Wireshark, I saw that the first REGISTER is send by my sip-phone to Kamailio. The first UNAUTHORIZED response came, as expected, with the WWW-Authenticate information. My phone sent a new REGISTER with the good realm and nonce-value (so I can see with wireshark) but Kamailio still refuse it by another UNAUTHORIZED message (same as the first one).
Can you please explain to me where am I wrong ?
If it helps, this is what I got on the kamailio logs :
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20466]: NOTICE: <core> [main.c:739]: handle_sigs(): Thank you for flying kamailio!!!
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20475]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20474]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20473]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20472]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20471]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20470]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20469]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20468]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[20467]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 kamailio: INFO: <core> [tcp_main.c:4746]: init_tcp(): using epoll_lt as the io watch method (auto detected)
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[22331]: INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[22331]: INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[22331]: INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 124928
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[22331]: INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 249856
Apr 21 14:27:39 fiducial-vm-centos66-sipr1 /usr/sbin/kamailio[22339]: INFO: ctl [io_listener.c:225]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config)
Hope someone could help me,
Thank you in advance anyway
Gab
Hi,
I'm using 'allow_trusted()' from Permissions module without a problem in a
Kamailio 4.0 routing logic. It does just work, but the MI command
'trusted_dump' tells me that the module is not in use:
'''
server:/home/jmillan# kamctl fifo trusted_dump
500 Trusted-module not in use
'''
Doing a 'trusted_reload' does neither make any query to database to
retrieve the data.
Other MI commands of same module do work correctly. ie: 'address_reload' or
'address_dump'
Note: I have tested in a Kamailio 1.5 and the result is the same.
Regards.
--
José Luis Millán
By port closed, I mean that ports are normally closed, but when rtpengine
send the first rtp packets to the client, it opens a pinhole in the
firewall, and the matching incoming packets from the client will make the
connection established,related in iptables. I think symmetric nat permits
that.
But now I'm thinking that it's impossible for rtpengine to know the
client's destination port at the learning phase if the client's rtp packets
can't reach rtpengine.
Rtpengine can learn the IP Address from kamailio through the --sip-source
CLI switch, but can't guess the port, right ?
So, playing with established,related is not possible.
> If the attacker is fast enough, yes. You can disable learning of
> endpoint addresses using the asynchronous flag, but obviously this will
> break NAT'd media. You can also use the strict-source flag to make
> rtpengine drop packets received from a mismatched source address.
So if I don't use strict-source flag, an attacker could merge any garbage
of data in an existing RTP stream ?
Thanks.
Hello,
it is time to nail down the roadmap to the next major release. We
discussed during the last IRC devel meeting, proposing to get it out by
beginning of June. Given we need to have at least one month of testing,
I propose the next milestones:
- freezing the development: Wednesday, April 22, 2015
- if testing goes smooth, then branching 4.3 after about one month:
During the week starting May 18
- test more in beta phase, prepare packaging, etc. and release after 2-3
weeks: One of the days between June 4 and 11
Other suggestions or adjustments are welcome! Send them to mailing list
to discuss further.
Cheers,
Daniel
--
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