sr-users October 2014

sr-users@lists.kamailio.org

101 participants
186 discussions

by Marko Seidenglanz

Hello, We are trying to use Kamailio 4.2 with RTPEngine 3.3 behind NAT. Somehow SIP responses (200 OK) are not handled by Kamailio. The following INVITE is send to the receiving end (WebRTC Client): INVITE sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK865.9db0c023bec2f314e89ae0f18b78d2f0.0 Via: SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK644600e2 Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as68f0a26a To: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com> Contact: <sip:anonymous@10.240.215.73:5060;alias=146.148.113.245~5060~1> Call-ID: 368f6ec901a3552e3b37a70907634f4f@10.240.215.73:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Fri, 24 Oct 2014 08:38:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 738 P-hint: outbound ....... The client response looks like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK865.9db0c023bec2f314e89ae0f18b78d2f0.0;rport=5060;received=104.155.11.255 , SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK644600e2 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as68f0a26a To: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com >;tag=E2VMB39U5NYHCXLZREFGQV2DAXPCZXNQ6J6P Contact: <sip:evQzk4l62MdF3G7U6eudj@whtest3.24dial.com >;tag=E2VMB39U5NYHCXLZREFGQV2DAXPCZXNQ6J6P Call-ID: 368f6ec901a3552e3b37a70907634f4f@10.240.215.73:5060 CSeq: 102 INVITE Record-Route: <sip:104.155.11.255:5060;nat=yes;lr=on> Content-Type: application/sdp Content-Length: 938 ....... I have a xlog call as the first statement in kamailio routing script. I can see the request, which get's handled correctly but the response does not appear in kamailio log, though it arrives at the machine. I am totally helpless, what might be wrong. If we use kamailio on a public accessable machine everything works as expected. Maybe anyone might have had similiar problems using kamailio behind NAT. Kind regards, Marko

9 years, 6 months

RPM Based installs...

by Derrick Bradbury

Just a question for the RPM maintainers... Is there a reason for some of the modules (such as carrierroute) not being included in the RPMS? It's been bugging me for a while, when I do an installation, I have to go and compile all the sources .... Not that it's an issue, just easier to do it from RPM if available. Thanks! Derrick

9 years, 6 months

Kamailio Infront of Asterisk with remote PBX

by Kenny Watson

Hi, I have a few asterisk servers providing some basic SIP trunking and routing. We have remote PBXs trunked onto asterisk which calls come into asterisk and are routing down to extensions on the remote PBX via prefix routing. I'm looking to have a central Kamailio Registrar/Proxy/Loadbalancer which Invites come into and are routed out to either SIP phones which are registered or to the remote PBX. I'm looking for some advice as to which modules would be best to use to achieve this as the remote PBXs will be dynamically registered rather than fixed gateways. Please let me know what further information would be helpful. Thanks Kenny Watson

9 years, 6 months

async workers

by Alex Balashov

Hi, What is the practical limit to the number of async worker processes? With SIP child processes, it seems to be about the number of available CPUs in /proc/cpuinfo. After that--at least, per my testing--one begins to hit the point of diminishing returns, presumably due to SHM IPC and synchronisation issues. Is the restriction similar in the async execution context? -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

9 years, 6 months

Crash after async_sleep().

by Vitaliy Aleksandrov

Hello list. root@proxy:/# kamcmd core.version kamailio 4.1.4 (i386/linux) I'm getting a crash when I'm trying to simulate a ringing UAS from kamailio config. To achieve this I've added the next config actions: > if (is_method("INVITE") && !has_totag() && $rU =~ "^999") { > sl_send_reply("100", "Fake Trying"); > sl_send_reply("180", "Fake Ringing"); > > t_set_fr(120000, 120000); > t_set_retr(65000, 65000); > t_set_auto_inv_100(0); > > async_sleep("60"); > send_reply("480", "Fake Temporary Unavailable"); > > t_release(); #<--- > > exit(); > }; I expect to receive 180 reply immediately and 480 after 60 seconds. The problem appears when a caller cancels a call after 60 seconds which is equal to async_sleep() wait time. I suppose there is a race condition happens between UDP worker that processes CANCEL request and timer process which calls t_continue() (tm's function). Calling t_newtran() explicitly doesn't change the situation. The only way I found to avoid crash is to remove t_release() call. In this case kamailio complains that "script write didn't release transaction" but continues to process new calls. With -DTIMER_DEBUG and -DDBG_QM_MALLOC I got next information: /usr/sbin/kamailio[11307]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0xb1d8c520), called from tm: h_table.c: free_cell(178), first free tm: h_table.c: free_cell(157) - aborting Crash mostly happens because of broken circular list of timers (timer_ln->next == NULL), but one time I saw got abort() during CANCEL request processing when tm tried to call free_cell() for an already freed pointer.

9 years, 6 months

Re: [SR-Users] One sided or no voice issue with websockets

by Amit Patkar

Please check ICE server settings. Your browser may be publishing local IP. One way voice or no voice is typical case when client is behind firewall. Regards, Amit Patkar dodul <dodul(a)live.com> wrote: >Hi > >I didn't get any responses from anyone regarding my issue. Can someone please give me some clue what I can do? Or if more information is needed please let know so that I can provide. > > >Sent from my Samsung Galaxy smartphone. > > >-------- Original message -------- >From: Kamrul Khan <dodul(a)live.com> >Date:10-21-2014 17:45 (GMT-06:00) >To: sr-users(a)lists.sip-router.org >Cc: >Subject: [SR-Users] One sided or no voice issue with websockets > >Hi, > >We have a setup with sipml5 to kamailio. It works perfectly within local network. In public network the signaling establishes perfectly, but most of the time we hear no voice, sometimes we hear one sided voice and in rare cases we hear voice from both sides. To fix this issue we configured our nathandler like the below: But, still no luck. Any idea how to fix this? Please HELP!!! > >modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") >modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") >. >. >. >route { >. >. > if (nat_uac_test("115")) { > if(nat_uac_test("64")) > force_rport(); > } > if (method=="REGISTER") { > fix_nated_register(); > add_rcv_param(); > } else { > fix_nated_contact(); > if(nat_uac_test("64")){ > if (!add_contact_alias()) { > xlog("L_ERR", "Error aliasing contact <$ct>\n"); > sl_send_reply("400", "Bad Request"); > exit; > } > } else { > add_rcv_param(); > } > } > } >. >. > >onreply_route { > if (nat_uac_test(64)) { > add_contact_alias(); > } >. >. >} > > > > >_______________________________________________ >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 > >_______________________________________________ >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

9 years, 6 months

Kamailio behind NAT (Asterisk - Kamailio - RTPEngine - Browser)

by Marko Seidenglanz

Hello, I have a problem with the following configuration. I want to make calls from Asterisk to a browser using RTPEngine as relay. Everything works fine, if Kamailio is not natted (See kamailio_without_nat.log). If it's address is translated, then 200 OK responses from the browser don't seem to be received by kamailio. I have configured Kamailio (4.2) and RTPEngine (3.3) both with advertisement of the public IP. Asterisk has public IP: 146.148.113.245 Kamilio has public IP: 104.155.11.255 Browser has public IP: 79.241.195.106 The INVITE after SDP Rewrite looks like this: INVITE sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255;lr=on;nat=yes> Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK683.5e75aa8b7f88561a91033a9b611fc0aa.0 Via: SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK5a79a7cf Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as53a7de72 To: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com> Contact: <sip:anonymous@10.240.215.73:5060;alias=146.148.113.245~5060~1> Call-ID: 523c5fda707c565c51b78c586247818e@10.240.215.73:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 23 Oct 2014 15:29:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 738 P-hint: outbound v=0 o=root 985629145 985629145 IN IP4 104.155.11.255 s=Asterisk PBX 12.5.0 c=IN IP4 104.155.11.255 t=0 0 a=ice-lite m=audio 31630 RTP/SAVPF 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:31631 a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:H9uLHKvstgjd58cZKa0LeRaxvf5XNJhaHgxDut7L a=setup:actpass a=fingerprint:sha-1 D9:26:45:E1:D6:E7:01:A4:04:90:F2:15:AF:A3:AD:01:3C:39:9B:7D a=ice-ufrag:YoTvWbLM a=ice-pwd:eifInSr3Oh8ZvMXjlE89mh8kF1OC a=tDndidate:krhMgLYHbAxWidfK 1 UDP 2130706431 104.155.11.255 31630 typ host a=candidate:krhMgLYHbAxWidfK 2 UDP 2130706430 104.155.11.255 31631 typ host The 200 OK like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK683.5e75aa8b7f88561a91033a9b611fc0aa.0;rport=5060;received=104.155.11.255 , SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK5a79a7cf From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as53a7de72 To: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com >;tag=6QTSN6N2SBW2FW49TMGJQJYZMGHQZ7LCUMQT Contact: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com >;tag=6QTSN6N2SBW2FW49TMGJQJYZMGHQZ7LCUMQT Call-ID: 523c5fda707c565c51b78c586247818e@10.240.215.73:5060 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 938 v=0 o=- 1907956949290984340 2 IN IP4 127.0.0.1 s=- t=0 0 m=audio 43590 UDP/TLS/RTP/SAVPF 0 8 101 c=IN IP4 79.241.195.106 a=rtcp:1 IN IP4 0.0.0.0 a=candidate:290309024 1 udp 2122260223 192.168.35.78 43590 typ host generation 0 a=candidate:2416297236 1 udp 1686052607 79.241.195.106 43590 typ srflx raddr 192.168.35.78 rport 43590 generation 0 a=candidate:1607352144 1 tcp 1518280447 192.168.35.78 0 typ host tcptype active generation 0 a=candidate:2416297236 1 udp 1686052607 79.241.195.106 43590 typ srflx raddr 192.168.35.78 rport 43590 generation 0 a=ice-ufrag:ZtbtZieUG9l22iCb a=ice-pwd:kRkmd0XuFdj+CKXVpInmK3yV a=fingerprint:sha-256 ED:DB:B8:D5:4D:38:1F:81:DC:94:9F:EB:6D:07:56:75:57:45:F4:F7:57:4D:C6:89:70:CC:13:6D:35:C0:8B:45 a=setup:active a=mid:audio a=sendrecv a=rtcp-mux a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=msid-semantic: WMS mPbtogqqqsv8DygpyOmxrTkjgW9aGzkbP3Vk Unfortunately Kamailio log does not tell me, that the response get's processed. (See kamailio_behind_nat.log). I traced with TShark and can see that the SIP/200 OK arrive at kamailio host. Any Idea, what might be missing in my routing script? The attachements can be found here: https://drive.google.com/folderview?id=0BxwAyaFvy_7fdHVnWXdFa2hEeHM&usp=sha… I would be very grateful for any help. Regards, Marko

9 years, 6 months

Kamailio behind NAT (Asterisk - Kamailio - RTPEngine - Browser)

by Marko Seidenglanz

Hello, I have a problem with the following configuration. I want to make calls from Asterisk to a browser using RTPEngine as relay. Everything works fine, if Kamailio is not natted (See kamailio_without_nat.log). If it's address is translated, then 200 OK responses from the browser don't seem to be received by kamailio. I have configured Kamailio (4.2) and RTPEngine (3.3) both with advertisement of the public IP. Asterisk has public IP: 146.148.113.245 Kamilio has public IP: 104.155.11.255 Browser has public IP: 79.241.195.106 The INVITE after SDP Rewrite looks like this: INVITE sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com SIP/2.0 Record-Route: <sip:104.155.11.255;lr=on;nat=yes> Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK683.5e75aa8b7f88561a91033a9b611fc0aa.0 Via: SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK5a79a7cf Max-Forwards: 69 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as53a7de72 To: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com> Contact: <sip:anonymous@10.240.215.73:5060;alias=146.148.113.245~5060~1> Call-ID: 523c5fda707c565c51b78c586247818e@10.240.215.73:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 12.5.0 Date: Thu, 23 Oct 2014 15:29:17 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 738 P-hint: outbound v=0 o=root 985629145 985629145 IN IP4 104.155.11.255 s=Asterisk PBX 12.5.0 c=IN IP4 104.155.11.255 t=0 0 a=ice-lite m=audio 31630 RTP/SAVPF 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=rtcp:31631 a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:H9uLHKvstgjd58cZKa0LeRaxvf5XNJhaHgxDut7L a=setup:actpass a=fingerprint:sha-1 D9:26:45:E1:D6:E7:01:A4:04:90:F2:15:AF:A3:AD:01:3C:39:9B:7D a=ice-ufrag:YoTvWbLM a=ice-pwd:eifInSr3Oh8ZvMXjlE89mh8kF1OC a=tDndidate:krhMgLYHbAxWidfK 1 UDP 2130706431 104.155.11.255 31630 typ host a=candidate:krhMgLYHbAxWidfK 2 UDP 2130706430 104.155.11.255 31631 typ host The 200 OK like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 104.155.11.255:5060 ;branch=z9hG4bK683.5e75aa8b7f88561a91033a9b611fc0aa.0;rport=5060;received=104.155.11.255 , SIP/2.0/UDP 10.240.215.73:5060 ;rport=5060;received=146.148.113.245;branch=z9hG4bK5a79a7cf From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as53a7de72 To: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com >;tag=6QTSN6N2SBW2FW49TMGJQJYZMGHQZ7LCUMQT Contact: <sip:eIh66yyxjlWNvNcuKWskH@whtest3.24dial.com >;tag=6QTSN6N2SBW2FW49TMGJQJYZMGHQZ7LCUMQT Call-ID: 523c5fda707c565c51b78c586247818e@10.240.215.73:5060 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 938 v=0 o=- 1907956949290984340 2 IN IP4 127.0.0.1 s=- t=0 0 m=audio 43590 UDP/TLS/RTP/SAVPF 0 8 101 c=IN IP4 79.241.195.106 a=rtcp:1 IN IP4 0.0.0.0 a=candidate:290309024 1 udp 2122260223 192.168.35.78 43590 typ host generation 0 a=candidate:2416297236 1 udp 1686052607 79.241.195.106 43590 typ srflx raddr 192.168.35.78 rport 43590 generation 0 a=candidate:1607352144 1 tcp 1518280447 192.168.35.78 0 typ host tcptype active generation 0 a=candidate:2416297236 1 udp 1686052607 79.241.195.106 43590 typ srflx raddr 192.168.35.78 rport 43590 generation 0 a=ice-ufrag:ZtbtZieUG9l22iCb a=ice-pwd:kRkmd0XuFdj+CKXVpInmK3yV a=fingerprint:sha-256 ED:DB:B8:D5:4D:38:1F:81:DC:94:9F:EB:6D:07:56:75:57:45:F4:F7:57:4D:C6:89:70:CC:13:6D:35:C0:8B:45 a=setup:active a=mid:audio a=sendrecv a=rtcp-mux a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=msid-semantic: WMS mPbtogqqqsv8DygpyOmxrTkjgW9aGzkbP3Vk Unfortunately Kamailio log does not tell me, that the response get's processed. (See kamailio_behind_nat.log). I traced with TShark and can see that the SIP/200 OK arrive at kamailio host. Any Idea, what might be missing in my routing script? I would be very grateful for any help. Regards, Marko

9 years, 6 months

RPC uac.reg_info hangs second time

by Dan Christian Bogos

Hey Guys, I am in the process of testing the reloading of the uac registrations (thanks Daniel for that), and hit some issue: * when using uac_reg_info twice in a row, second command always hangs. """ root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_info l_uuid 301 { l_uuid: 301 l_username: 301 l_domain: pbx.example.com r_username: 301 r_domain: pbx.example.com realm: asterisk auth_username: 301 auth_password: check123 auth_proxy: sip:172.16.254.75 expires: 360 flags: 0 diff_expires: -1414052722 timer_expires: 0 } root@Dev2:/etc/kamailio# root@Dev2:/etc/kamailio# root@Dev2:/etc/kamailio# root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_info l_uuid 301 ^C root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_info l_uuid 301 """ Do you see anything wrong with my data or did I hit a bug? Since we are here and could not find much documentation around, is there a way to decrease the default 150 seconds blocking reloads from the database? """ root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_reload root@Dev2:/etc/kamailio# kamcmd -s tcp:127.0.0.1:2046 uac.reg_reload error: 500 - Failed to shift records - check log messages """ Syslog: """ Oct 23 10:32:17 Dev2 /usr/sbin/kamailio[2693]: ERROR: uac [uac_reg.c:369]: uac_reg_ht_shift(): shifting the memory table is not possible in less than 150 """ Thanks in advance for any kind of tip! DanB

9 years, 6 months

Unix timestamp with millisecond precision

by Grant Bagdasarian

Hello, I'm currently using the $TS psuedovariable to get the current unix timestamp, but this only returns the timestamp up to a second precision. Is it possible in kamailio to get the current unix timestamp with milliseconds precision? Regards, Grant

9 years, 7 months

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sr-users October 2014