sr-users October 2014

sr-users@lists.kamailio.org

101 participants
188 discussions

Kamailio, authentication and MongoDB
by Nolan Darilek 24 Oct '14

24 Oct '14

Hello. I'm attempting to set up Kamailio for the first time and have a few questions. 1. The auth_db module specifically calls out MySQL, Postgres and flat file databases, but makes no mention of the new MongoDB module. Is the new MongoDB module supported as well and the docs just haven't been updated to account for that? 2. In looking at the auth_db parameters, I see values for column names, but nothing for a table/collection. How do I specify what table or collection my user data is in? 3. How do I create HA1 strings? The docs state: HA1 strings are MD5 hashes of username, password, and realm. Is that: md5(username+":"+password+":"+realm) or something else? Thanks.

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dispatcher behind NAT - lost ACK
by Nicholas Gill 24 Oct '14

24 Oct '14

Hello sr-users, We have Kamailio (behind NAT) configured acting as a proxy in front of some FreeSWITCH servers. There appears to be something amiss with my inbound (dispatcher) configuration which leads to misaddressing / misdelivery of the ACK after 200 OK (outbound calls appear to be proxied correctly [1]). Calls from the sip provider incoming to Kamailio are distributed by the dispatcher module to the FS server (For testing in this simplified scenario FS and Kamailio are on the same machine): SIP provider (5060) sends an invite to Kamailio(5061), Kamailio uses the dispatcher module to select a backend server and forwards the INVITE to FS(5070) (see inbound-callflow.png; kamailio.ext and kamailio.int are the same machine, just public/private addresses). I notice that at no point is a Via/record-route header for the FS server inserted into the forwarded session. I'm not actually certain this is a requirement, but I can't think of another obvious way that Kamailio could proxy the same session to the same FS server. The 100 Trying seems unremarkable [2], I suspect the 200 OK [3] is problematic. It has been proxied from the FS server, however contains no reference to the FS server address either in the via headers nor record-route (kamailio.int:5070). The incoming ACK [4] then appears to be misdelivered / lost - Kamailio receives it on the private address and forwards it to the public address rather than the FS server. My configuration [5] is built based on the default configuration + examples from the dispatcher module. There are some provisions for FreeSWITCH internal/external profiles made so the configuration listens on 2 different ports. This particular scenario should only use the 5061 port as it involves calls to an external sip provider (briefly 5060 should be proxied to FS:5080 and 5061 should be proxied to FS:5070 and vice-versa). If someone can see an issue with the configuration and/or point to an error in the call flow (i.e. should FS be inserting the Via header?) that would be greatly appreciated. Thanks, -nick [1] Outbound call flow (see also outbound-callflow.png) FS(port 5070) sends an invite to Kamailio(5061) (Kamailio is configured in FS as an outbound proxy), INVITE contains Via header for the FS server, Kamailio forwards to sip provider, and routes all messages back and forth correctly. [2] 100 Trying SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.0;rport=5060 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com> Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 INVITE Server: kamailio (4.1.6 (x86_64/linux)) Content-Length: 0 [3] Proxied 200 OK (Kamailio -> Sip Provider) SIP/2.0 200 OK Via: SIP/2.0/UDP sip.provider.com:5060;rport=5060;branch=z9hG4bKfffb.7c81ee53.0 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK4Qg7Ng27BvHrK Record-Route: <sip:kamailio.ext:5061;lr=on> Record-Route: <sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr;did=0ec.84ff82b1> From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com>;tag=BFvQmggHrg74m Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 INVITE Contact: <sip:0390156842@kamailio.int:5070;transport=udp> User-Agent: IMX Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-FS-Support: update_display,send_info [4] Incoming ACK (Sip Provider -> Kamailio) ACK sip:0390156842@kamailio.ext:5061;transport=udp SIP/2.0 Record-Route: <sip:sip.provider.com;lr;ftag=j3KQmpvmg6mvr> Via: SIP/2.0/UDP sip.provider.com:5060;branch=z9hG4bKfffb.7c81ee53.2 Via: SIP/2.0/UDP far.external.ip;received=far.external.ip;rport=5060;branch=z9hG4bK509ZQBKB947aF Route: <sip:kamailio.ext:5061;lr=on> Max-Forwards: 69 From: "Nicholas Gill" <sip:0384171966@far.external.ip>;tag=j3KQmpvmg6mvr To: <sip:0390156842@sip.provider.com>;tag=BFvQmggHrg74m Call-ID: aa7174e7-d028-1232-4b95-001cc0dd11e9 CSeq: 66413636 ACK Contact: <sip:gw+Exetel@far.external.ip:5060;transport=udp;gw=Exetel> Proxy-Authorization: Digest username="0384171900", realm="far.external.ip", nonce="xxxx", algorithm=MD5, uri="sip:0390156842@sip.provider.com", response="xxxx" Content-Length: 0 P-hint: rr-enforced [5] Kamailio configuration (based on default configuration). Dispatcher set 1 is FS internal profile (5080) and set 2 is external profile (5070) ... #!define WITH_NAT ... alias="kamailio.int:5060" alias="kamailio.int:5061" alias="kamailio.ext:5060" alias="kamailio.ext:5061" listen=udp:kamailio.int:5060 advertise kamailio.ext:5060 listen=udp:kamailio.int:5061 advertise kamailio.ext:5061 port=5060 port=5061 ... loadmodule "dispatcher.so" modparam("dispatcher", "flags", 2) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #modparam("dispatcher", "ds_ping_interval", 60) ... modparam("nathelper", "sipping_from", "sip:pinger@etellicom.com") ... request_route { ... # Not from FS servers gets dispatched to FS servers. if(!ds_is_from_list("1") and !ds_is_from_list("2")) { route(DISPATCH); } # Requests from FS External profile routed out via specific port. if(ds_is_from_list("2")) $fs="udp:kamailio.int:5061"; ... } route[DISPATCH] { if(dst_port==5061) { $fs="udp:kamailio.int:5061"; # Hash over to uri for test purposes if(!ds_select_dst("2", "2")) { send_reply("404", "No destination"); exit; } } else { # Hash over to uri for test purposes if(!ds_select_dst("1", "2")) { send_reply("404", "No destination"); exit; } } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } ...

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sip response not handled by kamailio
by Marko Seidenglanz 24 Oct '14

24 Oct '14

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

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RPM Based installs...
by Derrick Bradbury 24 Oct '14

24 Oct '14

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

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Kamailio Infront of Asterisk with remote PBX
by Kenny Watson 24 Oct '14

24 Oct '14

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

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async workers
by Alex Balashov 24 Oct '14

24 Oct '14

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

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Crash after async_sleep().
by Vitaliy Aleksandrov 24 Oct '14

24 Oct '14

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.

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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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sr-users October 2014