sr-users September 2014

sr-users@lists.kamailio.org

96 participants
171 discussions

R: Re: R: Re: RTPPROXY & BRANCH
by Marino Mileti 30 Sep '14

30 Sep '14

Unfortunately rtpengine doesn't work in this way. At the end of the calls this is the output log: Final packet stats: Tag 'Fw3D7R0', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30224 <> 192.168.10.20:7078 , 540 p, 92880 b, 0 e Media #1, port 30225 <> 192.168.10.20:7079 (RTCP), 3 p, 324 b, 0 e Media #2, port 30256 <> 192.168.10.20:9078 , 0 p, 0 b, 0 e Media #2, port 30257 <> 192.168.10.20:9079 (RTCP), 3 p, 264 b, 0 e Tag 'qWE6Gsh', created 0:41 ago, in dialogue with 'TTPyT~Hdw' Media #1, port 30140 <> 192.168.10.50:7078 , 533 p, 91068 b, 0 e Media #1, port 30141 <> 192.168.10.50:7079 (RTCP), 5 p, 444 b, 0 e Media #2, port 30170 <> 192.168.10.50:9078 , 0 p, 0 b, 0 e Media #2, port 30171 <> 192.168.10.50:9079 (RTCP), 1 p, 88 b, 0 e Tag 'TTPyT~Hdw', created 0:41 ago, in dialogue with 'Fw3D7R0' Media #1, port 30206 <> 172.20.11.208:7078 , 1070 p, 183736 b, 0 e Media #1, port 30207 <> 172.20.11.208:7079 (RTCP), 4 p, 496 b, 0 e Media #2, port 30240 <> 172.20.11.208:9078 , 4188 p, 1435946 b, 0 e Media #2, port 30241 <> 172.20.11.208:9079 (RTCP), 4 p, 400 b, 0 e 192.168.10.x clients are natted..it seems that rtpengine open 2 ports (for example video) for each receiver (30256 & 30170) and 1 port for the caller (30240). But on the INVITE of Kamailio only video port 30170 is offered to receivers, instead on caller side there are 2 distinct 183s message that offer 30190 & 30240. It's a little bit strange because some of these port doesn't appear in the log of rtpengine. At the end I can see video only on one receiver I don't know if the problem is on Kamailio (rtpproxy-ng module) or in the rtpengine :) > Without rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted > client so no needs of rtpproxy) > - B offers port b1,b2 (183) > - C offers port c1,c2 (182). > - A starts to send audio/video RTP to B on port b1,b2 > - A starts to send audio/video RTP to C on port c1,c2 > > With rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to Kamailio.... > - Kamailio contact rtpproxy because B&C are natted clients > - rtpproxy check callid and offer offers port k1,k2.... > - Kamailio sends INVITE to B offering k1,k2 > - Kamailio sends INVITE to C offering k1,k2 > - B offers port b1,b2 (183) > - C offers port c1,c2 (182) > - Kamailio sends 183 to A (for B leg) offering p1,p2 > - Kamailio sends 183 to A (for B leg) offering p3,p4 > - A starts to stream on p1,p2,p3,p4 but only one receiver can see the video > (B or C depends who will be the first:)) > > I don't know if it depends on that B & C receives same ports; i don't know > if rtpproxy is able to "duplicate" stream received from A to all "receiver" If A sends two streams, there is no need for duplication. A sending to p1 should be forwarded to B (b1) and A sending to p3 should be forwarded to C (c1). Both should be able to receive media sent by A. I believe that's what rtpengine does (but I haven't tested it). The reverse direction might be more confusing, but a final 200 OK with SDP should fix that. cheers _______________________________________________ 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

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Call Group versus MAX_BRANCHES limit
by João Vitor Arruda 30 Sep '14

30 Sep '14

Hi folks, I have a question related with the limited number of branches being 12 in config.h #define MAX_BRANCHES 12 /*!< maximum number of branches per transaction */ I am trying to implement a Call Group that consists in trying each member of the group (that can result in a parallel forking when lookup() is used) in sequence (similar to serial forking). Currently I have written code that is similar to the one described here: http://www.kamailio.org/dokuwiki/dokuphp/tutorials:avpops#serial_forking The only difference is that for each member I call lookup() and then in the failure_route(), I pull the next member from the stack and call lookup() again. The code works fine when used for groups with less than 12 members. However, call groups with 12 or more members fails as I can never reach remaining extensions once MAX_BRANCHES limit is reached. I've tried another approach using the functions described here: http://kamailio.org/docs/modules/4.1.x/modules/tm.html#tm.serial_forking but it too uses a new branch for every group member until the 12 limit is reached out. I also tried to use the functions "remove_branch(index)" and "clear_branches()" (both of which are poorly documented. In fact the only reference I ever found for these functions was here: http://www.kamailio.org/wiki/features/new-in-3.2.x#functions). Unfortunately, I wasn't able to prevent the MAX_BRANCHES limit from being reached. Ultimately, my goal is to have a limitless Call Group. Do you have any suggestions? (other than increasing the hard coded MAX_BRANCHES limit) Thank you, Joao Arruda

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Re: [SR-Users] Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 30 Sep '14

30 Sep '14

Hi,I ran the code on my server (as Daniel suggested) and then tried to send message from one user to other but no traffic was detected.ngrep d any qt W byline "sip" port 5060interface: anyfilter: (ip or ip6) and ( port 5060 )match: sipAlso, after I tried to install the PRESENCE module and having restarted the systems today, I noticed that the response that I am getting on Jitsi is "403 Not Relaying" and not "403 Not Allowed". I am not sure what caused the change.I hope someone can help me figure it out.Cheers!Gaurav Original message From:"DanielConstantin Mierla"< miconda(a)gmail.com >Date: 30 Sep 14 16:19:30Subject: Re: [SRUsers] Kamailio with Jitsi Presence troubleTo: ; "Kamailio (SER) Users Mailing List"Hello,you should grab the SIP traffic on kamailio server and send it to the mailing list. In this way we can spot why you get the 403 Not Allowed. You can use:ngrep d any qt W byline "sip" port 5060Cheers, DanielOn 30/09/14 08:50, Gaurav Kumar wrote:Hello all!I tried sendin g this earlier too but since I was not subscribed tothe mailing list, it bounced back.I am trying to setup a secure videoconferencing setup for mysmall office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used it on aUbuntu 12.04 setup and want to setup a SIP server for very fewusers (less than 10, at most) to use it through Jitsi. I am tryingto follow the tutorials available at: http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://www… am able to log into Jitsi on independent machines, both behindand without a firewall. But the trouble I am facing is that theusers do not appear online to each other. If I try to send a textmessage nonetheless, I get the "403 Not allowed" error.At first, I tried the config file for Kamailio provided on thefirst link. It had some problems due to it being an old versionand the config file on the 2nd link help ed me out. I suspect thereis some problem with the PRESENCE module but I do not know whatsince it does not give any error whatsoever. I added the "#!defineWITHPRESENCE" line and installed the Presence module but to noavail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00bfor Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a weeknow. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now! SIP Express Router (SER) and Kamailio (OpenSER) srusers mailing list srusers(a)lists.siprouter.org http://lists.siprouter.org/cgibin/mailman/listinfo/srusersDanielConstantin Mierla http://twitter.com/#!/miconda http://www.linkedin.com/in/micondaGet Yourself a cool, short @in.com Email ID now!

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Kamailio with Jitsi - Presence trouble
by Gaurav Kumar 30 Sep '14

30 Sep '14

Hello all!I tried sending this earlier too but since I was not subscribed to the mailing list, it bounced back.I am trying to setup a secure videoconferencing setup formy small office. After a lot of convincing, my bosses have allowed meto create a setup and have given me a live IP. I have used it on aUbuntu 12.04 setup and want to setup a SIP server for very few users(less than 10, at most) to use it through Jitsi. I am trying to followthe tutorials available at:http://kb.asipto.com/kamailio:skypelikeserviceinlessthanonehour&https://… able to log into Jitsi on independent machines, both behind andwithout a firewall. But the trouble I am facing is that the users do notappear online to each other. If I try to send a text messagenonetheless, I get the "403 Not allowed" error.At first, I triedthe config file for Kamailio provided on the first link. It had someproblems due to it being an old version and the config fil e on the 2ndlink helped me out. I suspect there is some problem with the PRESENCEmodule but I do not know what since it does not give any errorwhatsoever. I added the "#!define WITHPRESENCE" line and installed thePresence module but to no avail. My current kamailio.cfg file can be seen at: http://pastebin.com/bZJxVLfL(I have hidden my live IP in the text).My current /etc/kamailio/kamctlrc file can be seen at: http://pastebin.com/tV7Z9E8eI can upload the logs/other file content as needed. I am a n00b for Kamailio so you will have to be patient with me.Please help me out here. I have been after it for almost a week now. Cheers!GauravDear srusers! Get Yourself a cool, short @in.com Email ID now!

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Uri transformations stopped working after upgrade from 3.3 to 4.1.6.
by Julia Boudniatsky 30 Sep '14

30 Sep '14

Hello, Some Uri transformations stopped working after upgrade from 3.3 to 4.1.6. *To: <sip:01412127775678;cic=012;csel=noind@10.25.153.150:5060 <http://noind@10.25.153.150:5060>;user=phone>* *In v3.3* $tu({uri.params}) return: cic=012;csel=noind $tu({uri.param,cic}) return: 012 *In v4.1.6* $tu({uri.params}) return : user=phone $tu({uri.param,cic}) return : Best regards, Julia

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Re: [SR-Users] Questions about multiple contacts in kamailio
by Federico Cabiddu 30 Sep '14

30 Sep '14

Hi, you can achieve this by using the flag 0x04 when calling save() function from registrar module: http://kamailio.org/docs/modules/stable/modules/registrar.html#idp1965144 Regards, Federico On 30 Sep 2014 10:54, "qw" <applemax82(a)163.com> wrote: Hi, I have one question about contact in kamailio. Sometimes, one user may register multiple contacts in kamailio server, where contacts are represented as ip:port. For example, the user registers at first, and lose connection to the internet later. After one minute, the user connects the internet again, but can't get service from kamailio server. Then the user need to registers with new contact. Now there are two contacts in kamailio, i.e. old one and new one. How can I remove the old contact, and make sure there is only one contact in kamailio server. Looking forward to your help! B.R. andrew _______________________________________________ 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

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Questions about multiple contacts in kamailio
by qw 30 Sep '14

30 Sep '14

Hi, I have one question about contact in kamailio. Sometimes, one user may register multiple contacts in kamailio server, where contacts are represented as ip:port. For example, the user registers at first, and lose connection to the internet later. After one minute, the user connects the internet again, but can't get service from kamailio server. Then the user need to registers with new contact. Now there are two contacts in kamailio, i.e. old one and new one. How can I remove the old contact, and make sure there is only one contact in kamailio server. Looking forward to your help! B.R. andrew

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Kamailio -PSTN Gateway
by Zaka Ul Isam 30 Sep '14

30 Sep '14

Hello Folks: Please help with above, I have browsed and tried various suggestions on this list without much luck! I think problem can be reduced to three questions ;) 1: Apart from DEFINE WITH PSTN directive, do I need certain modules to be compiled? (DIALPLAN, CARRIERROUTE, LCR aso) If yes, then we ought to put up a list of modules required for each task/ activity..... 2: How Kamailio core decides the call routing? i.e: based on dialed digits as specified in dialplan . # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00355)[0][4-9]{3,20}$")) return; && #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "10.1.6" desc "PSTN GW Address" pstn.gw_port = "5080" desc "PSTN GW Port" #!endif 3: If Kamailio is listening on say wlan0 interface 10.42.0.1:5060 (Ubuntu HOT SPOT) can it route to PSTN Gateway/ Softswitch on P2P1 (Eth0) without Bridging? Put another way, does kernel route towards the specified interface or Kamailio is capable of routing based on active routing cache? Thanking you in advance, Zak ________________________________ Albtelecom & Eagle Mobile invite you to protect together the Environment. Please do not print the e-mail unless really necessary. Albtelecom & Eagle Mobile iu ftojnë të mbrojmë së bashku Mjedisin. Lutemi të mos e printoni këtë komunikim elektronik nëse nuk është vërtet i nevojshëm. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the addressee/s. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited. If you receive this e-mail in error, please notify the sender immediately and delete it! Albtelecom makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message may belong to sender alone and may not necessarily reflect the opinions of Albtelecom. Albtelecom shall bear no liability for any loss or damage caused by software or e-mail viruses. Ky mesazh dhe çdo informacion i transmetuar në përmbajtje te këtij mesazhi është konfidencial dhe është i destinuar vetëm për marrësin e destinuar. Nëse nuk jeni marrësi i destinuar, Ju bëjmë me dije se çdo përhapje, transmetim, kopjim apo përdorim i çdo informacioni është i ndaluar. Nëse e merrni këtë mesazh gabimisht, ju lutem kontaktoni urgjentisht nisësin e tij dhe fshijeni atë. Albtelecom nuk jep asnjë garanci për saktësinë apo plotësinë e informacionit në përmbajtje të këtij mesazhi dhe nuk mban asnjë përgjegjësi për informacionin e përmbajtur, transmetimin, marrjen, ruajtjen apo përdorimin e tij në çfarëdolloj mënyre. Mendimet e shprehura në këtë mesazh mund t'i përkasin vetë nisësit dhe nuk mund të reflektojnë domosdoshmërisht qëndrimet e Albtelecom. Albtelecom nuk do të mbaje asnjë përgjegjësi për humbje ose dëme te shkaktuara nga programet apo viruset.

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kamailio call limit
by Yuriy Gorlichenko 30 Sep '14

30 Sep '14

Hello. I try to test with SIPp my stak of kamailio->asterisk. I run SIPp with 200 calls/sec and see only 68 at maximum active calls at server. When I set 500 calls/sec with limit 1000 I see 68 active connections again. So when I try test SIPp to asterisk without Kam i see wright maximum of active channels/calls. My question - does Kamailio have limits ffor calls? If no- help me to understand please what is going on with my server. Thanks.

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Re: [SR-Users] R: Re: R: Re: RTPPROXY & BRANCH
by Richard Fuchs 29 Sep '14

29 Sep '14

On 09/29/14 14:51, Marino Mileti wrote: > Without rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to B,C (in case of no natted > client so no needs of rtpproxy) > - B offers port b1,b2 (183) > - C offers port c1,c2 (182). > - A starts to send audio/video RTP to B on port b1,b2 > - A starts to send audio/video RTP to C on port c1,c2 > > With rtpproxy: > > - A offers port a1,a2 (audio video) in INVITE to Kamailio.... > - Kamailio contact rtpproxy because B&C are natted clients > - rtpproxy check callid and offer offers port k1,k2.... > - Kamailio sends INVITE to B offering k1,k2 > - Kamailio sends INVITE to C offering k1,k2 > - B offers port b1,b2 (183) > - C offers port c1,c2 (182) > - Kamailio sends 183 to A (for B leg) offering p1,p2 > - Kamailio sends 183 to A (for B leg) offering p3,p4 > - A starts to stream on p1,p2,p3,p4 but only one receiver can see the video > (B or C depends who will be the first:)) > > I don't know if it depends on that B & C receives same ports; i don't know > if rtpproxy is able to "duplicate" stream received from A to all "receiver" If A sends two streams, there is no need for duplication. A sending to p1 should be forwarded to B (b1) and A sending to p3 should be forwarded to C (c1). Both should be able to receive media sent by A. I believe that's what rtpengine does (but I haven't tested it). The reverse direction might be more confusing, but a final 200 OK with SDP should fix that. cheers

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sr-users September 2014