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37572 discussions

Kamailio v4.1.3 Released
by Daniel-Constantin Mierla 25 Apr '14

25 Apr '14

Hello, Kamailio SIP Server v4.1.3 stable release is out. This is a maintenance release of the latest stable branch, 4.1, that includes fixes since release of v4.1.2. There is no change to database schema or configuration language structure that you have to do on installations of v4.1.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.1.3. For more details about version 4.1.3 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2014/04/kamailio-v4-1-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Caller ID fields incorrect on Polycom 650s
by J Lee 24 Apr '14

24 Apr '14

Hi there. I have a situation where the caller id information on my phone appear incorrectly. When I have ext 1 dial ext 2, on the caller's phone, instead of saying "To: 2", it says "From: 2". I'm using kamailio version 4.0.4. I'm just new to everything kamailio and VoIP so I apologize in advance for any remedial questions. So far, I've done a SIP trace to and checked all packets line by line to see if I could spot where the To address was erroneously changed to FROM. In particular, I was checking the call-info fields for caller id data or any other data what would appear on the LED of the phone. Now I'm wondering, who's job (aka which module) is it to set the caller id information for both phones? I'm thinking of digging inside the code for the right module just to see if I can follow what's going on In addition to answering the question above, if you have any suggestions on what else I can check or try, I'd appreciate it. Thanks List!

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Problem accessing git repository
by arun Jayaprakash 24 Apr '14

24 Apr '14

Is there a problem with the git repository? When I try to download kamailio (git clone --depth 1 git://git.sip-router.org/sip-router kamailio ) I get an error saying that git.sip-router.org can not be reached. Please let me know, thank you. Arun

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Re: [SR-Users] TCP reset
by Ciprus, Daniel 24 Apr '14

24 Apr '14

Understand :-) thank you Sent via the Samsung GALAXY S®4, an AT&T 4G LTE smartphone -------- Original message -------- From: Carsten Bock Date:04/24/2014 6:09 AM (GMT-05:00) To: "Ciprus, Daniel" Cc: "Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] TCP reset Hi Daniel, Define "near future"... :-) Once we have time, we may look at this topic; but no timeline yet.... Kind regards, Carsten 2014-04-23 15:55 GMT+02:00 Daniel Ciprus <daniel.ciprus(a)acision.com>: > Thanks Carsten, > > Any plans to do so in near future ? :-) > > On 04/23/2014 09:52 AM, Carsten Bock wrote: > > Hi Daniel, > > what you want, is basically already implemented in the "Standard" > usrloc module, see: > > http://kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.handle_… > > We haven't implemented that feature yet in the IMS modules.... > > Schöne Grüße, > Carsten > > > 2014-04-23 15:47 GMT+02:00 Daniel Ciprus <daniel.ciprus(a)acision.com>: > > Hi, > > Is there any way to remove registration entry from registrar (Kamailio IMS > built on stable 4.1) detected on p-cscf by incoming TCP RST from the network > ? What's happening is that client is loosing connections/crashing and TCP > RST is sent back to pcscf which is not propagating changes back to > registrar. This is likely not possible with my version of kamailio but at > least logging which would indicate change of the status will be helpful. > > thanks for any hints > > > -- > Daniel Ciprus > Integration engineer > http://www.acision.com > > 9954 Mayland Dr > Suite 3100 > Richmond, VA 23233 > USA > T: +1 804 762 5601 > E: daniel.ciprus(a)acision.com > > ________________________________ > This e-mail and any attachment is for authorised use by the intended > recipient(s) only. It may contain proprietary material, confidential > information and/or be subject to legal privilege. It should not be copied, > disclosed to, retained or used by, any other party. If you are not an > intended recipient then please promptly delete this e-mail and any > attachment and all copies and inform the sender. Thank you for > understanding. > > > _______________________________________________ > 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 > > > > -- > Daniel Ciprus > Integration engineer > http://www.acision.com > > 9954 Mayland Dr > Suite 3100 > Richmond, VA 23233 > USA > T: +1 804 762 5601 > E: daniel.ciprus(a)acision.com > > ________________________________ > This e-mail and any attachment is for authorised use by the intended > recipient(s) only. It may contain proprietary material, confidential > information and/or be subject to legal privilege. It should not be copied, > disclosed to, retained or used by, any other party. If you are not an > intended recipient then please promptly delete this e-mail and any > attachment and all copies and inform the sender. Thank you for > understanding. > -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/ ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

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TCP reset
by Daniel Ciprus 24 Apr '14

24 Apr '14

Hi, Is there any way to remove registration entry from registrar (Kamailio IMS built on stable 4.1) detected on p-cscf by incoming TCP RST from the network ? What's happening is that client is loosing connections/crashing and TCP RST is sent back to pcscf which is not propagating changes back to registrar. This is likely not possible with my version of kamailio but at least logging which would indicate change of the status will be helpful. thanks for any hints -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

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Planning release of v4.1.3
by Daniel-Constantin Mierla 24 Apr '14

24 Apr '14

Hello, I am considering releasing v4.1.3 by mid of next week, on Wednesday or Thursday (April 23 or 24). If there are issues you are aware of and not reported to the bug tracker, add them there asap to investigate them. Also, if you noticed some fixes in the master branch not backported yet, report them to the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Difference between route(RELAY) and t_relay
by aawaise 24 Apr '14

24 Apr '14

Hi all, I want to know while coding kamailio.cfg. What factor determines that route(RELAY) should be used or t_relay ?? Thanks. Cheers. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Difference-between-route-RELAY-and-… Sent from the Users mailing list archive at Nabble.com.

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LDAP and Kerberos backend for authentification / Or PAM / SASL ?
by Yoann Gini 23 Apr '14

23 Apr '14

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

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IBM SIP server has implemented proxy managed overload protection
by Yang Hong 23 Apr '14

23 Apr '14

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

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rtpproxy_manage() issues in rtpproxy-ng
by Alex Balashov 23 Apr '14

23 Apr '14

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? -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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