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37549 discussions

Re: [SR-Users] One sided or no voice issue with websockets
by Amit Patkar 23 Oct '14

23 Oct '14

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

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Kamailio behind NAT (Asterisk - Kamailio - RTPEngine - Browser)
by Marko Seidenglanz 23 Oct '14

23 Oct '14

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

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Kamailio behind NAT (Asterisk - Kamailio - RTPEngine - Browser)
by Marko Seidenglanz 23 Oct '14

23 Oct '14

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

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RPC uac.reg_info hangs second time
by Dan Christian Bogos 23 Oct '14

23 Oct '14

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

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Unix timestamp with millisecond precision
by Grant Bagdasarian 23 Oct '14

23 Oct '14

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

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hello. I'm a user for kamailio. I wanna ask something
by 신희준 23 Oct '14

23 Oct '14

Hello I have a question about uac module. When i use uac_req_send function, from tag is made by this function automatically. I wanna make a PRACK request which is in-dialog request, so i don't want to append new from-tag at that situation. I think there some solution, but i can't find.. can you help me?? thank you

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usrloc: Unregister client on TCP connection close
by Camille Oudot 22 Oct '14

22 Oct '14

Hi, I'm looking for a mechanism to unregister clients when the corresponding TCP connection is closed, but had no success so far. I've tried the handle_lost_tcp parameter of usrloc, but it didn't work: the user was not unregistered, whereas the connection was seen as closed by kamailio. Is it still a relevant option? Otherwise, is there any mechanism that would trigger some kind of event_route whenever a connection (TCP, TLS, WS, WSS) is lost or closed? Cheers, -- Camille

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Re: [SR-Users] Connecting IMS and Webrtc!!
by Carsten Bock 22 Oct '14

22 Oct '14

Hi Dionlar, can you elaborate the Memory issue a little bit more? Can you also send me more information, on what OS and what Kamailio-Version you are using? What configuration are you using? If you have a memory issue, feel free to contact sr-users(a)lists.sip-router.org, they should be able to help you as well. Probably no-one of our team (core-networks@) will be able to help you further, as i am taking care of the implementation myself. Thanks, Carsten 2014-10-22 1:27 GMT+02:00 dionlar lang <dionlar.lang(a)gmail.com>: > Hello dear!! > > I am Lang, undergraduate student research at the University Sheik Anta Diop > of Dakar and I have a memory issue focusing on the interconnection between > IMS and Webrtc.Je am writing to ask your proposals because I just read some > of your articles about the subject.Is it you please help me. > > Thank you and I count on you! -- 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 5247593-0 Fax +49 40 5247593-99 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/

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Re: [SR-Users] One sided or no voice issue with websockets
by dodul 22 Oct '14

22 Oct '14

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

1 0

Asynchronous sql operation also for acc module?
by Sebastian Damm 22 Oct '14

22 Oct '14

Hi, I read the article about asynchronous sqlops: http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-4-asynchronous-sql.h… I was wondering whether there are any plans of bringing that feature to the acc module, too. We have seen it in the past that a blocking database server brought our proxy to a halt because the acc module was blocking, too. So this could help a lot if the acc entries were written asynchronously. Best Regards, Sebastian

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