Thank you all for the answers. Finally, there is not a very big problem
if one side is disconnected because the other side will end dialog when
phone will be hang up
have a good day :)
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Le -10/01/-28163 22:59, Reda Aouad a écrit :
Not also that the recommend session timer value that UACs generally
use by default is 1800 seconds, or 30 minutes. Just in case you may
accept it. To me, it's not acceptable when customers are billed.
However, the risk is not so big. If one of the UACs is suddenly
disconnected, the other party will most likely hang up, the dialog is
terminated and finally we loose only a few seconds of billing.
Reda
On Tue, Mar 20, 2012 at 10:02, Reda Aouad <reda.ao...@gmail.com
<mailto:reda.ao...@gmail.com>> wrote:
Hello,
In SIP, session timers can be used to periodically ping the UAC
(using a re-INVITE or UPDATE) to know if it's alive or not. Then
action can be taken - terminating the call.
Kamailio has the SST (SIP Session Timer) module which only
enforces a minimum session timer value for UACs, but not a maximum
one. It doesn't ping the UACs neither. This is fine because the
RFC stops here. A nice improvement to Kamailio would be to augment
the SST module with a feature which enforces a maximum session
timer value and pings UACs. Another suggestion would be to rely on
nathelper's keepalive results to take a decision after a keepalive
times out, but then we'd have to terminate all dialogs in which
the UAC that is not responding is present, since nathelper's
keepalive are out-of-dialog. No very neat, but functional.
And I don't think the dialog module can do anything about this
problem.
I know that what I am suggesting may not be defined in RFCs, and
so are some features of SIP servers, but in my opinion should be
implemented as it adds a great value to Kamailio.
We cannot rely on RTP timeout since a UAC may use a
silence-detection codec and be silent for some time, or may put a
call on hold for a while, not sending RTP packets in both cases.
This is why RTP timeout detection is not reliable. Anyway,
mediaproxy timeouts ONLY AND ONLY in the case it doesn't receive
RTP packets from BOTH UACs, not only one, for the reasons
mentioned. I don't know about rtppoxy, maybe others can tell more
about it.
One solution if you really need to solve your problem would be to
put a B2BUA in the SIP path, such as Asterisk or FreeSwitch. They
enforce a maximum session timer which UACs can use to ping
themselves every now and then, and Asterisk can even ping the UACs
and terminate the call if one of them doesn't respond. The
downside: lower performance and higher cost. Asterisk is very
heavy and Kamailio can handle many, many more calls, so you'll
have to load balance to several Asterisk servers if you have a
single Kamailio machine handling thousands of simultaneous calls.
Kamailio developers out there, what about boosting the SST module
with new features? Or creating an SSTX module?
Reda
On Tue, Mar 20, 2012 at 07:35, SamyGo <govoi...@gmail.com
<mailto:govoi...@gmail.com>> wrote:
Hi,
Yes that is the behaviour when the media isn't flowing through
a regulatory tool (in-terms it sees the media and know call is
actually going on rtpproxy/media-proxy) but in the absence of
any such tool SIP server is not aware that the call-media is
still in progress or is dead ! so it always assume that the
call is active and hence the BYE signals are never originated
from server end to shutdown the call.
I am definitely not an expert but I am guessing
that dialogue module do some keepalive tests for an ongoing
session and not sure what it do if either end fails to respond !!
Regards,
Sammy
On Tue, Mar 20, 2012 at 11:04 AM, Vineet Menon
<mvineetme...@gmail.com <mailto:mvineetme...@gmail.com>> wrote:
i guess it should time out...the other end...since it has
no way of knowing that the other end is no more present...
Regards,
Vineet Menon
On 20 March 2012 11:30, Rabary <te...@gulfsat.mg
<mailto:te...@gulfsat.mg>> wrote:
Hi mailing,
Newbie to kamailio, I follow this tuto
http://nil.uniza.sk/sip/kamailio/adding-mysql-support-kamailio-31-debian-lenny
for the registration SIP via mysql database and it
works fine, but I saw that when during the call we
disconnect the called UAC from network or turn the
power off the caller UAC don't hangup.
Is there any tool for how to hangup call when the UAC
on the other side has no network connection or it
isn't power on durring a call ?
I heard for mediaroxy or rtpproxy but I don't know if
them can do what I except to haveand we also use ip
routing to make kamailio server to communicate with
the UAC so we don't use NAT.
Our topology is:
kamailio (with public IP address) ---> cisco switch
---> LAN ---> UAC (with private IP address)
Thanks in advance.
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