sr-users February 2015

sr-users@lists.kamailio.org

95 participants
130 discussions

RTPEngine - No support for kernel packet forwarding available
by Muhammad Shahzad 05 Feb '15

05 Feb '15

Hi, I have latest stable release of RTPEngine deployed in a virtual machine (KVM) along with Kamailio v4.2. All is working fine except i see this message in RTPEngine logs, -- rtpengine[16455]: [82qrjq0hdtt45afbqo98 port 40960] Kernelizing media stream rtpengine[16455]: [82qrjq0hdtt45afbqo98 port 40960] No support for kernel packet forwarding available -- using lsmod command i see that module xt_RTPENGINE is load. -- Module Size Used by iptable_mangle 12488 0 xt_RTPENGINE 22068 3 iptable_nat 12800 0 nf_nat 17924 1 iptable_nat ... -- Also the iptables rule for RTPEngine exists, i.e. -- Chain INPUT (policy ACCEPT 224K packets, 37M bytes) pkts bytes target prot opt in out source destination 224K 37M rtpengine all -- any any anywhere anywhere Chain FORWARD (policy ACCEPT 0 packets, 0 bytes) pkts bytes target prot opt in out source destination Chain OUTPUT (policy ACCEPT 224K packets, 37M bytes) pkts bytes target prot opt in out source destination Chain rtpengine (1 references) pkts bytes target prot opt in out source destination 2022 414K RTPENGINE udp -- any any anywhere anywhere RTPENGINE id:0 -- Yet still RTPEngine is not able to kernelize the RTP stream and does user-space packet forwarding? Can you guys suggest what could be wrong here? Thank you.

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Issues while using Kamailio as proxy with Asterisk
by Juned Khan 05 Feb '15

05 Feb '15

Hi All, I am using Kamailio as proxy server with Asterisk but having some issues with Inbound and outbound calls. Asterisk is a registrar, Kamailio is a just for proxy. Using softphone call are working fine but using hardphone, its creating issues. Here are the test cases. No Test case *98 Inbound 1 ATA registered with Proxy Domain No No 2 ATA registered with Proxy IP Yes Yes 3 ATA registered with Asterisk domain No No 4 ATA registered with Asterisk IP Yes Yes 5 Soft phone registered with Proxy Domain Yes Yes 6 Soft phone registered with Proxy IP Yes Yes 7 Soft phone registered with Asterisk domain Yes Yes 8 Soft phone registered with Asterisk IP Yes Yes Here I am dialling *98 with the mentioned scenario, if it goes to voicemail options then Yes, else No I am using ALLO ATA. Here is my http://pastie.org/9887806 Regards Juned Khan / Network Engineer +91 9974740823/ juned.khan(a)inextrix.com iNextrix Technologies Pvt Ltd. www.inextrix.com <http://htmlsig.com/www.inextrix.com> [image: Facebook] <https://www.facebook.com/junedkhan23> [image: Twitter] <https://twitter.com/juned23> [image: Google Plus] <https://plus.google.com/+JunedKhan> [image: Linkedin] <http://htmlsig.com/in.linkedin.com/in/junedk> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company.

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Send sip INFO req from kamailio
by Uri Shacked 04 Feb '15

04 Feb '15

Hi, Any idea how to send a SIP INFO req ? I am familiar with uac_req_send. but how do I send it with in a specific dialog and with data in the INFO req ? BR, Uri

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kamailio 4.2.1 core in timer.c:877
by 张顺通 03 Feb '15

03 Feb '15

I user rtimer module : loadmodule "rtimer.so" modparam("rtimer", "timer", "name=p_route;interval=1;mode=1;") modparam("rtimer", "exec", "timer=p_route;route=P_ROUTE") application will coredump in sometimes, and core file in below, what's the reason? Program terminated with signal 11, Segmentation fault. #0 0x00000000005fda98 in timer_list_expire (t=1043586690, h=0x7f9ed45ac9b0, slow_l=0x7f9ed45adc80, slow_mark=2316) at timer.c:877 877 timer.c: No such file or directory. in timer.c (gdb) bt full #0 0x00000000005fda98 in timer_list_expire (t=1043586690, h=0x7f9ed45ac9b0, slow_l=0x7f9ed45adc80, slow_mark=2316) at timer.c:877 tl = 0x7f9ed47f7e98 ret = 0 #1 0x00000000005fc686 in timer_handler () at timer.c:953 saved_ticks = 1043586690 run_slow_timer = 0 i = 268 __FUNCTION__ = "timer_handler" #2 0x00000000005fdd4d in timer_main () at timer.c:992 No locals. #3 0x00000000004aa571 in main_loop () at main.c:1700 i = 12 pid = 0 si = 0x0 si_desc = "udp receiver child=11 sock=10.1.*.*:5160\000\000\b\000\000\000\000\000\000\000\000\060S\324\236\177\000\000\060=#z\001\000\000\000(\205V\324\236\177\000\000\060=#z\377\177\000\000\214\216N\000\000\000\000\000\037t\372D\000\000\000\000(\205V\324\236\177\000\000\220@ #z\377\177\000\000\000\000\000\000\001\000\000" nrprocs = 12 __FUNCTION__ = "main_loop" #4 0x00000000004af166 in main (argc=11, argv=0x7fff7a234098) at main.c:2561 cfg_stream = 0x1da75e0 c = -1 r = 0 tmp = 0x7fff7a234681 "" tmp_len = 32767 port = 2049130512 proto = 32673 options = 0x706ec0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 220012340 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0xb p = 0x0 __FUNCTION__ = "main"

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[OT] ssl certificates updated for kamailio.org
by Daniel-Constantin Mierla 03 Feb '15

03 Feb '15

Hello, short note to say that I updated the ssl certificates for kamailio.org web site. They are issued by cacert.org, which is probably not trusted by most of the browsers, but they are well known in the open source environment. If you get a warning about certificate change and the issuer is cacert.org, then you should be safe to login over https (e.g., to edit the wiki pages). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Issues while using Kamailio as proxy with Asterisk
by Juned Khan 03 Feb '15

03 Feb '15

Hi All, I am using Kamailio as proxy server with Asterisk but having some issues with Inbound and outbound calls. Here are the test cases. No Test case *98 Inbound 1 ATA registered with Proxy Domain No No 2 ATA registered with Proxy IP Yes Yes 3 ATA registered with Asterisk domain No No 4 ATA registered with Asterisk IP Yes Yes 5 Soft phone registered with Proxy Domain Yes Yes 6 Soft phone registered with Proxy IP Yes Yes 7 Soft phone registered with Asterisk domain Yes Yes 8 Soft phone registered with Asterisk IP Yes Yes Here I am dialling *98 with the mentioned scenario, if it goes to voicemail options then Yes, else No I am using ALLO ATA. I have attached my kamailio.cfg with this email. Regards Juned Khan / Network Engineer +91 9974740823/ juned.khan(a)inextrix.com iNextrix Technologies Pvt Ltd. www.inextrix.com <http://htmlsig.com/www.inextrix.com> [image: Facebook] <https://www.facebook.com/junedkhan23> [image: Twitter] <https://twitter.com/juned23> [image: Google Plus] <https://plus.google.com/+JunedKhan> [image: Linkedin] <http://htmlsig.com/in.linkedin.com/in/junedk> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. E-mails are not secure and cannot be guaranteed to be error free as they can be intercepted, amended, or contain viruses. Anyone who communicates with us by e-mail is deemed to have accepted these risks. Company Name is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company.

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Ro_CCR function
by Shane Harrison 03 Feb '15

03 Feb '15

Hi all, In a previous post I was alerted to the (perhaps obvious) fact that that ims_charging module is design to run within an IMS environment: http://lists.sip-router.org/pipermail/sr-users/2015-January/086675.html I have subsequently attempted to modify the module to work with the standard usrloc Kamailio module. I decided to branch from the 4.2.2 release rather than the master since I was unsure of the state of the ims_charging module in the master. It now compiles fine and installs. I also have the cdp module talking watchdog_requests to a freeDiameter server so at that level things seem OK. However inserting the Ro_CCR() call into the kamailio cfg file shows up some problems, primarily: Feb 2 01:05:52 hh-rcs-sipproxy3 /usr/sbin/kamailio[6473]: WARNING: usrloc [dlist.c:624]: register_udomain(): Registering a new domain called 'voip.we-rcs.flowcloud.systems' with usrloc Feb 2 01:05:52 hh-rcs-sipproxy3 /usr/sbin/kamailio[6473]: ERROR: <core> [db.c:443]: db_check_table_version(): invalid version 0 for table voip.we-rcs.flowcloud.systems found, expected 6 (check table structure and table "version") Exploring this error leads me to a few questions: 1) The 4.2.2 Ro_CCR() has the diameter domain as a parameter. The addition of this function to my kamailio.cfg file causes ro_fixup() -> domain_fixup() -> ul.register_domain which as per the log above, attempts to look for a table with the same name as the domain. Is it expected that such a table exists, for standard Kamailio I thought all usrloc records simply go into the Location table , regardless of the domain. 2) What is the purpose of the ro_fixup() 3) I notice that the 4.3 module documentation for ims_charging module has 6 parameters for the function call while the example code given in the same section as only 5 (missing domain). The code from the master branch for the function is different again - not sure if 4.3 is still in quite a bit of flux or not for the ims_charging module? 4) Should I be using 4.3 ims_charging module as a starting point or is 4.2.2 good enough i.e are there major changes? At this stage I am going to create a table for the domain but would like some clarity on how the usrloc and domains relate to table names. Kind regards Shane Shane Harrison Senior Software Engineer Imagination Technologies NZ Limited Level 2 1 Market Grove Lower Hutt, 5010 New Zealand PO Box 30-449 Lower Hutt, 5040 New Zealand Phone: +64 4 890-3681 ext 3361

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Grouped dispatching
by Grant Bagdasarian 02 Feb '15

02 Feb '15

Hello, Would the below scenario be possible using the dispatcher module or any other routing module? Destination 1: 192.168.1.1 Destination 2: 192.168.1.2 Destination 3: 172.168.1.1 These three destinations belong to the same distribution group(set), but Kamailio has to do the following: Perform round-robin using destination 1 and 2, but if both are down fail-over to destination 3. I hope I'm being clear enough. Regards, Grant

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Video Key-Frame Request using RTCP FIR or SIP INFO message
by Muhammad Shahzad 02 Feb '15

02 Feb '15

Hi, This may be a bit out of focus topic for this forum but i am posting it here anyway with hope that some guru would shed some light on it and point me to right direction. The problem is that i want to establish video call between a webrtc and a sip client using kamailio (for signalling) and RTPEngine (for media relay). Both signalling and the audio stream seems to work perfectly fine The remote video on webrtc client side (i.e. video stream from sip client) takes about 20-30 seconds to establish but once it starts it works fine. However, the remote video on sip client side (i.e. video stream from webrtc client) starts almost immediately (within 3-5 seconds) but it gets stuck after 1 or 2 seconds, then it goes blank after about 30 seconds. After a long discussion with sip client developer, we now understand the fact that sip client sends a request for so called key-frame, which is ignored by webrtc client. This request is sent through both RTCP stream and SIP INFO message. The SIP INFO message seems to be pointless as media is internally managed by chrome/firefox and these browsers don't give us such sophisticated access and control over media streams. Please let me know if this assumption is wrong. For the RTCP stream based request (RTCP-FIR), i only see "Invalid RTCP packet type" error message in RTPEngine logs (not sure if it drops this packet or relay it anyway). Does anyone has any idea on how can we either, 1. Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO message and issue a key-frame in RTP video stream in response to this SIP request? OR 2. Force RTPEngine to accept RTCP-FIR and issue key-frame in RTP video stream on webrtc client's behalf? If there is any other solution to this, please feel free to share. Thank you.

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What could cause an intermittent fault (28) with http_query()?
by Tim Chubb 01 Feb '15

01 Feb '15

Hi All I seem to be experiencing an intermittent fault with the utils http_query() method. We are implementing a routing and white list component that is accessed via a REST api, however i have observed several occasions where this is logged: Jan 30 16:04:17 vs-kam-prod02 /usr/local/sbin/kamailio[13184]: ERROR: utils [functions.c:149]: http_query(): failed to perform curl (28) The indicated error number (28) seems to suggest a timeout is occurring with curl, however examining a capture of network traffic when this happens shows that a http request is not sent from the server to the destination at all, usually attempting a second call results in everything working correctly. As stated its intermittent in its nature and so I cannot reliably trigger this issue, other than through sheer repetition, so any ideas as to what could be causing this issue would be gratefully received,

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sr-users February 2015