On Wednesday, October 21, 2015, at 17:00GMT (12:00CT, 18:00 London,
19:00 Berlin), the Cluecon weekly conference call will focus on Kamailio
and FreeSwitch.
I will be joining the call, answering the questions about Kamailio and
its options to integrate with FreeSwitch. Expect the FreeSwitch core
developers to be around to handle the questions about their project.
Participation is open for anyone, you can dial in for audio or video
sessions using a SIP phone or webrtc capable browser:
- sip:888@conference.freeswitch.org
- https://cantina.freeswitch.org/vc
More dial in options (e.g., PSTN) are presented at:
-
https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere…
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.com
i noticed that entries that i had added to htable at sip proxy start in
event_route [htable:mod-init], e.g.,
$sht(htable=>app_srv::sbc::<null>) = "sip:SBC:SEMS_PORT";
had disappeared from htable after some (don't know exactly how long)
time. i have not specified any autoexpire for that table:
modparam("htable", "htable", "htable=>size=HTABLE_SIZE;dbtable=htable;")
after restart of sip proxy they reappeared in htable like this:
# sip-proxy_ctl htable.get htable 'app_srv::sbc::<null>'
{
item: {
name: app_srv::sbc::<null>
value: sip:127.0.0.1:5090
flags: 2
expire: NEVER
}
}
what could explain the disappearance?
-- juha
Hi guys,
What do you think about the RFC 5393 on loop detection and amplification
attack protection?
The RFC is short and still a proposed standard but don't you think it could be useful to prevent loop and amplification attack? Because even if the max-forward field reduces the loop to ~70 hosts (in most cases) with some techniques we could fork the message up to 2^70 messages (as described in the RFC) to crash the servers.
Basically the server has to do 2 things:
* check if it is not already in the via of the message
* the previous check is not enough as a B2BUA could have replace the via headers, so the RFC introduces a new field called max-breadth to limit the forking.
I have not seen a lot of implementation of this RFC on the free SIP software and I think it could be a good way to improve kamailio making a module for it (the easier way to implement this feature I think).
In fact I'm in a research internship about VoIP security and
I have time to develop such a module for kamailio if you think it's a
good idea (I'm looking for some security improvements in free software solutions so if you have other idea don't hesitate to tell me).
Cheers,
Tetram
Hi All,
I've come accross an intermittent issue where an initial publish is sent
to our presence server, proxy recieves the subsequent 200 with etag, but
the following publish sent does not contain the sip-if-match header of
the recieved 200, which ends up causing trouble with BLF on our test
devices, it also has the added side affect of not being cleared out of
the presentity table, these entries need to be manually removed.
This does not occur all the time, I can use the same phone and make
several calls before this issue occurs. It is not limited to a device
type, we've tried with panasonic, polycom, cisco, zoiper, yealink and
linksys devices, all exhibit the same sporadic behaviour.
I've attached an example sip trace of the publish messages where this
happens, you can see the second publish doesnt have the sip-if-match
header whose content should be that of the sip-etag in the previous 200
to the initial publish.
I'm not sure if there is something I am misunderstanding here, a
configuration issue, or if this is indeed a bug.
Both the proxy and the presence server are running kamailio v4.3.3.
Any pointers/suggestions/guidance would be appreciated.
If you need any additional info, please dont hesitate to ask.
Thanks
Hello,
Is it possible to have Kamailio send a ReINVITE every X minutes to determine if a session is still active?
I know it's a proxy and doesn't have B2BUA capabilities, but there was a module which allowed Kamailio to generate SIP messages, but I can't find it anymore.
If Kamailio is not the place to do this, which component in the voip network should be responsible for this? Are there perhaps other ways to poll for session state in Kamailio?
Regards,
Grant
Both boxes are running Debian 8.
I’m still looking at the mysql logs, but I haven’t found anything relevant, yet.
> On Oct 14, 2015, at 5:00 AM, sr-users-request(a)lists.sip-router.org wrote:
>
> Hello,
>
> do you have centos+selinux? Because I heard about a lot of limits with
> the combination of the two.
>
> Have you checked the mysqld syslog messages, anything relevant there?
>
> Cheers,
> Daniel
>
Dear all,
I have installed Kamailio and Rtpproxy. Following is the process (ps) info:
"kamailio 648 1 0 Oct08 ? 00:00:05 /usr/bin/rtpproxy -s
udp:127.0.0.1 7722 -u kamailio kamailio -p /var/run/rtpproxy/rtpproxy.pid -l
192.168.58.25"
But when I give command: " kamctl mi nh_show_rtpp" it returns following:
"500 command 'nh_show_rtpp' not available"
What could be the problem?
Thanks,
Amar.
