Hello,
My users are managed on a common LDAP / Kerberos setup. I don’t have access in any ways to the user password in clear text mode.
I’ve start to follow this tutorial http://www.kamailio.org/wiki/tutorials/mini-howto-admin/ldap-user-auth and check this documenation http://kamailio.org/docs/modules/4.1.x/modules/ldap.html
But it seems LDAP module provided in Kamailio don’t handle this scenario, it except to have user password in clear text mode in the LDAP database (wrong in so many ways…).
Do we have a way with existing Kamailio plugin to use a authentication backend who don’t provide access to clear text password?
I provide different authentication « adapter » who can help any service not compatible with Kerberos to handle authentication:
- ldap bind
- sasl
- pam
Do we’ve a way to use one of this backend as an authentication service (and not just a users database).
Best regards,
Yoann
Hello Folks.
According the new update on Feb 5, 2014, IBM has implemented proxy managed overload protection (PMOP) at SIP proxy server.
http://pic.dhe.ibm.com/infocenter/wasinfo/v7r0/index.jsp?topic=%2Fcom.ibm.w…
The above IBM link lists IBM SIP proxy server custom properties for SIP overload protection/control.
•burstResetFactor
•deflatorRatio
•dropOverloadPackets
•inDialogAveragingPeriod
•maxThroughputFactor
•outDialogAveragingPeriod
•perSecondBurstFactor
•proxyTransitionPeriod
•sipProxyStartupDelay
IBM SIP overload protection mechanism can be regarded as "Local SIP Overload Protection/Control" (Proceedings of WWIC, June 2013).
I am moderator of open-source FreeRDP-WebConnect project at GitHub.
https://github.com/FreeRDP/FreeRDP-WebConnect/issues/36
I will try to integrate two implicit SIP overload protection/control algorithms (RRRC, IEEE Globecom 2010 and RTDC, IEEE ICC 2011) into Kamailio (OpenSER) in the future.
A survey on SIP overload protection/control algorithms (including IETF RFC "SIP Overload Control") can be downloaded from the following ResearchGate link.
Y. Hong, C. Huang, and J. Yan, “A Comparative Study of SIP Overload Control Algorithms,"Network and Traffic Engineering in Emerging Distributed Computing Applications, Edited by J. Abawajy, M. Pathan, M. Rahman, A.K. Pathan, and M.M. Deris, IGI Global, 2012, pp. 1-20.
http://www.researchgate.net/publication/231609451_A_Comparative_Study_of_SI…http://arxiv.org/abs/1210.1505
This survey paper provides a short review on SIP Express Router (SER). As we know, SIP Express Router (SER) and Kamailio (OpenSER) are open-source SIP router.
"SIP Express Router (SER) provides a load balancing module to mitigate the overload caused by large subscriber populations or abnormal operational conditions (IP Telecommunications Portal, 2011).
http://www.igi-global.com/chapter/comparative-study-sip-overload-control/67…
The presentation slides for two implicit SIP overload protection/control algorithms (RRRC and RTDC) are available for your download.
Redundant Retransmission Ratio Control (RRRC) - implicit SIP overload protection/control algorithm (IEEE Globecom 2010 Slides) can be downloaded from the following ResearchGate link.
https://www.researchgate.net/publication/258555827_Mitigating_SIP_Overload_…
Round-Trip Delay Control (RTDC) - implicit SIP overload protection/control algorithm (IEEE ICC 2011 Slides) can be downloaded from the following ResearchGate link.
https://www.researchgate.net/publication/257945199_Round-Trip_Delay_Control…
The paper with Redundant Retransmission Ratio Control (RRRC, implicit SIP overload protection/control) algorithm can be downloaded from the following ResearchGate link.
Y. Hong, C. Huang, and J. Yan, "Mitigating SIP Overload Using a Control-Theoretic Approach," Proceedings of IEEE Globecom, Miami, FL, U.S.A, December 2010.
https://www.researchgate.net/publication/221284946_Miigating_SIP_Overload_U…http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5683124
Redundant Retransmission Ratio Control (RRRC, implicit SIP overload protection/control) algorithm has been quickly adopted by The Central Weather Bureau of Taiwan for their early earthquake warning system.
Ting-Yun Chi, Chun-Hao Chen, Han-Chieh Chao, and Sy-Yen Kuo, "An Efficient Earthquake Early Warning Message Delivery Algorithm Using an in Time Control-Theoretic Approach", 2011.
http://link.springer.com/chapter/10.1007%2F978-3-642-23641-9_15#http://www.ipv6.org.tw/docu/elearning8_2011/1010004798p_3-7.pdf
Short review and comments on RRRC implicit SIP overload protection/control algorithm by former IEEE TAC Associate Editor S. Mascolo:
Luca De Cicco, Giuseppe Cofano, and Saverio Mascolo, "Local SIP Overload Control", Proceedings of WWIC, June 2013.
http://link.springer.com/chapter/10.1007%2F978-3-642-38401-1_16#http://c3lab.poliba.it/images/2/2a/SipOverload_WWIC13.pdf
The paper with Round-Trip Delay Control (RTDC, implicit SIP overload protection/control) algorithm can be downloaded from the following ResearchGate link.
Y. Hong, C. Huang, and J. Yan, "Design Of A PI Rate Controller For Mitigating SIP Overload," Proceedings of IEEE ICC, Kyoto, Japan, June 2011.
https://www.researchgate.net/publication/224249824_Design_of_a_PI_Rate_Cont…http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5963029
Round-Trip Delay Control (RTDC, implicit SIP overload protection/control) algorithm has been recommended as White Paper by TechRepublic (CBS Interactive)
http://www.techrepublic.com/whitepapers/design-of-a-pi-rate-controller-for-…
Control theoretic approaches have been applied to model the interactions between an overloaded SIP server and its upstream servers as a feedback control system in two different scenarios - Round-Trip Delay Control (IEEE ICC 2011) and Redundant Retransmission Ratio Control (IEEE Globecom 2010).
Journal paper (implicit SIP Overload Protection/Control) published by Springer Telecommunication Systems not only conducts more theoretical analysis of Round-Trip Delay Control (RTDC) and Redundant Retransmission Ratio Control (RRRC), but also discusses how to apply RTDC algorithm to mitigate SIP overload for both SIP over UDP and SIP over TCP (with TLS).
Journal paper can be downloaded from the following ResearchGate link.
Y. Hong, C. Huang, and J. Yan, "Applying control theoretic approach to mitigate SIP overload", Telecommunication Systems, Volume 54, Issue 4, December 2013, pp 387-404.
https://www.researchgate.net/publication/257667871_Applying_control_theoret…http://link.springer.com/article/10.1007/s11235-013-9744-8
Implicit SIP overload protection/control solution using control theoretic approaches can be regarded as an engineering applications of control theory, see the discussion on control system design in the answers to the following two ResearchGate(RG) questions. ResearchGate is the largest social network for research scientists and engineers in the world.
RG Question #1: "What are trends in control theory and its applications in physical systems (from a research point of view)?"
https://www.researchgate.net/post/What_are_trends_in_control_theory_and_its…
RG Question #2: "What are the latest problems to be solved in "control of nonlinear systems"?"
https://www.researchgate.net/post/What_are_the_latest_problems_to_be_solved…
RG Question #3: "What are the new trends in control engineering?"
https://www.researchgate.net/post/What_are_the_new_trends_in_control_engine…
Best regards,
Winston Hong
Software Engineer
Ottawa, Ontario
Canada
Hello,
I'm running the latest pull of Kamailio 4.1 (last commit
be187e135b0b9b28136817c3569ab5c0abcc5b3f) and am using rtpproxy-ng with
a recent mediaproxy-ng master (commit
cb6990e43864b077dd6a24acfbdf5ef76c1a427e).
For no apparent reason, Kamailio has stopped sending 'offer' commands to
it when I use rtpproxy_manage(). My use of it is in this setting:
if(isflagset(PROXY_MEDIA) && !isflagset(PROXY_MEDIA_SET) &&
has_body("application/sdp")) {
set_rtp_proxy_set("1");
rtpproxy_manage("o");
add_rr_param(";proxy_media=yes");
setflag(PROXY_MEDIA_SET);
}
I put in a log message to confirm that this block is being run. It is.
But, there's no offer command going to the mediaproxy-ng on the other
box, as confirmed by packet captures and logs. However, the
mediaproxy-ng is getting lots of 'answer' and 'delete' commands that it
doesn't know what to do with, since the call was never initialised with
'offer'.
When I change rtpproxy_manage() to rtpproxy_offer(), it works perfectly.
I've been using rtpproxy_manage() forever, almost as long as it's
existed. Does it not work with rtpproxy-ng, despite what the
documentation says?
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Suite 106
Decatur, GA 30030
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Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
I'm having some troubles with a provider sending RTP before a 183 Session
Progress or 200 OK (I see up to 1s of rtp prematurely). The machine is running
rtpproxy and apparently rtpproxy buffers these rtp packets and flushed them in
one burst when the 183/200 arrive, this creates havock in some endpoints but
is undesirable in all cases IMHO if the delay of the 182/200 is to high
(>0.1s).
Is there a way to control buffering/flushing from kamailio? Is rebuilding
rtpproxy with a smaller buffer an option? Or should I switch to an other proxy
module? Any thoughts about this subject (apart from getting the provider to
stop sending premature rtp)?
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http://www.pocos.nl/ - http://www.sipo.nl/
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Hello all,
I'm trying to implement Kamailio 4.1 with Asterisk 12.1.0 regarging this
tutorial:
> http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
And when I try to compile kamaili.cfg, I still got this error:
ERROR route "FROMASTERISK" not found
I have loaded all modules like in tutorial.
I tried to find some solve this issue, but with no result.
Hi,
Or is it possible to use a regular expression for the user?
I basically want certain users to use a certain route without having to add
them all in.
Keith
Hi all
Following situation
1 dispatcher/loadbalancer getting all the inbound traffic and sending it to 3 different gateway (round robin).
The loadbalancer has no (or even very few) business logic.
Just "in" - split to different gateway - "out"
3 sip gateway doing all the business logic like auth, modifications on header, different checks, ...
These 3 gateways use the same database (cluster) with routing tables and so on.
A call gets terminated to a carrier through these sip gateway.
Now I would like to implement call limiting (no of calls) by using either only dialog module or cnxcc module based on "source ip" or later on "cli".
My problem:
A call from one source ip can be sent to the (3) different sip gateway - so not all calls are processed by the same sip gateway.
How can I ensure, that only a certain number of calls are allowed - even if they are split up on the 3 different gateway?
Or do I need to implement this kind of business logic on the dispatcher/loadbalancer?
That would not make much sense, because this is just a "stupid" machine...
Thanks for helping
Oli
Hello all,
I want to pick the domain name of a particular online user from location
list of siremis and then perform some action on this basis. How can it be
picked ??
Secondly how can the user name of called user be picked from the INVITE
packet received by the server ??
Any help will be appreciated.
Thanks,
Regards,
Aawaise.
--
View this message in context: http://sip-router.1086192.n5.nabble.com/Pick-domain-name-of-a-user-from-loc…
Sent from the Users mailing list archive at Nabble.com.
Hi,
I am trying to use the drouting module and was wondering if there is a
wildcard character for the dr_groups table for the user?
My situation is that I want to look at any user from a certain IP range
(i.e. the carrier).
Any ideas?
Thanks
Keith
Hi, i see this tutorial
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb,
I've done step by step as described in the tutorial, but i have a problem,
when A extension dial to B extension, B extension doesn't ringing...
I send my sip.conf and kamailio.cfg
Addittional, asterisk and kamailio are installed on same server, which have
one private IP (192.168.50.217) and public IP 200.41.110.94
--
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