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37517 discussions

Kamailio rtpengine sdp
by gerry kernan 21 Apr '17

21 Apr '17

Hi Thanks in advance if anyone can point me in the correct direction . I have kamailio running in front of some asterisk VM's. SIP endpoint register to their asterisk PBX via Kamailio dispatcher module. I'm running rtpengine with a Wan and private interface to bridge audio stream between these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio. Calls from ext to ext work fine audio both directions , calls outbound to PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both directions. But incoming calls via SIP provider I only get audio on stream from asterisk registered ext to external caller , no audio from external caller to the asterisk ext. I reckon I have something wrong in my Kamailio.cfg . if I register an ext direct to asterisk I get audio both ways on incoming calls. And rtp logs from rtpenegine show it as trying to send the rtp to the private address of the sip endpoint rather that its WAN address. I think my mistake in somewhere in the cfg below. Do I need to handle invites from the backend asterisk servers different that invites from sip endpoints? Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

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NOTIFY request from PCSCF is on TCP transport rather than UDP
by Haggerty, Paul (phaggert) 21 Apr '17

21 Apr '17

Hello, I've got an IMS client which uses UDP for transport. Our application server is also using UDP. We are using a Kamailio IMS Core. The client REGISTERS with UDP and also does a SUBSCRIBE. Our application server responds with a NOTIFY to the client. Looking at the trace. The messages between the PCSCF and SCSCF are TCP. The REGISTER and SUBSCRIBE responses to the client from the PCSCF are UDP (which I would expect). However, when the PCSCF get the NOTIFY is tries to open a TCP connection to the Client. (I assume to send a NOTIFY). Any ideas of what is happening here? I'm fairly new to the Kamailio IMS. I am more familiar with the Open IMS Core. (which is not having this issue). Thanks, Paul

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How ICE works with private IPs as candidates?
by Denys Pozniak 21 Apr '17

21 Apr '17

Hello! I have next topology: ISP->Kamailio + Rtpengine->WebRTC Sometimes I receive from Browser side SDP like below with private IPs in ICE candidates. - Please explain why it happens? - How ICE finds a pair? I think Rtpengine obtains packet from Browser and somehow matches with peer (as it can't reach Browser via private IP) U KAMAILIO_IP:34150 -> RTPENGINE_IP:22222 0_16586_545 d3:sdp962:v=0. o=- 3981552194926701250 2 IN IP4 127.0.0.1. s=-. t=0 0. a=msid-semantic: WMS 5efe361e-087d-47de-98f6-9b51427d3764. m=audio 50589 RTP/SAVPF 8 0. c=IN IP4 192.168.2.205. a=rtcp:9 IN IP4 0.0.0.0. a=candidate:710145796 1 udp 2122255103 <(212)%20225-5103> 2001::9d38:6abd:3c21:28f:3f57:fd32 50588 typ host generation 0 network-id 1 network-cost 50. a=candidate:2355154215 1 udp 2122194687 <(212)%20219-4687> 192.168.2.205 50589 typ host generation 0 network-id 2. a=ice-ufrag:sPAh. a=ice-pwd:RjXqOz8rpwjhKPU3BzXkJd1d. a=fingerprint:sha-256 04:AF:D4:6C:07:C2:DD:18:48:E0: BC:46:15:15:00:5F:C9:AD:CF:8D:F3:94:55:5A:54:0C:37:B3:E6:44:63:EC. a=setup:active. a=mid:audio. a=sendrecv. a=rtcp-mux. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=ssrc:3055599516 <(305)%20559-9516> cname:n0Cdw7SNbYDIba8s. a=ssrc:3055599516 <(305)%20559-9516> msid:5efe361e-087d-47de-98f6-9b51427d3764 41750b25-d640-4bad-8953-55f4e2684c49. a=ssrc:3055599516 <(305)%20559-9516> mslabel:5efe361e-087d-47de- 98f6-9b51427d3764. a=ssrc:3055599516 <(305)%20559-9516> label:41750b25-d640-4bad-8953- 55f4e2684c49. 3:ICE6:remove5:flagsl13:trust-address14:media-handover8: SDES-offe7:replacel6:origin18:session-connectione18: transport-protocol7:RTP/AVP8:rtcp-muxl5:offer6:accepte7: call-id37:domain.com_29465313:received-froml3:IP412:10.23. 255.93e8:from-tag13:QQcg5eFv08S9K6:to-tag10:nsebpmf7ia7:command6:answere

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http_client_query API of http_client module issue.
by vivek＠advaitamtech.com 21 Apr '17

21 Apr '17

Hi All, Am executing the curl GET request using the API http_client_query(url, [post-data], result) like its mentioned in the http_client module page. In the result part of the called API, I am receiving only the first line of the response to the GET request sent. How do I get or access the full response? Thanks, Vivek.

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Kamailio World 2017 - grants for students and underrepresented people
by Daniel-Constantin Mierla 21 Apr '17

21 Apr '17

Hello, with a bit of delay, given that had to accommodate and accept some requests from the last year queue, I am glad to announces that we can offer two more seats at Kamailio World Conference, May 8-10, 2017, in Berlin. Therefore we are continuing the program from last years, based on the roots and the tight relation of Kamailio project with the academic environment, the eligible people being students enrolled in universities or research institutes (both bachelor and PhD programs qualify) as well as people from underrepresented groups. If you think you are eligible and want to participate, email me directly (miconda(a)gmail.com) or to registration(a)kamailio.org . Participation to all the content of the event (workshops, conference and social event) is free, but you will have to take care of expenses for traveling and accommodation. Write a short description about your interest in real time communications and, when it is the case what is the university or the research institute you are affiliate to. Also, if you are not a student, but you are in touch with some or have access to students forums/mailing lists, it will be very appreciated if you forward these details. Time is short, we already received few requests based on my remark in a message sent out few weeks ago, therefore in a matter of days we will do the selection, so you have to hurry up a bit. More information about Kamailio World is available on the web site: - http://www.kamailioworld.com Many thanks to the event sponsors that allowed to continue this program, respectively: FhG Fokus, Asipto, Flowroute, Telnyx, Sipwise, Sipgate, Simwood, Obihai, NG-Voice, Evariste Systems, Digium, VoiceTel, Pascom and Core Network Dynamics. Expect a full house event! Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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New developer: Guillaume Bour
by Daniel-Constantin Mierla 21 Apr '17

21 Apr '17

Hello, I would like to announce that Guillaume Bour (https://github.com/gbour) has now developer privileges on Kamailio's gihub project. He has contributed recently a new module named keepalive: - http://kamailio.org/docs/modules/devel/modules/keepalive.html There is also a pull request from him waiting to be merge to drouting module, adding capability of detecting active/inactive gateways used by the module. Thanks for the contributions so far and looking forward to collaborate in the future! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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http_async_query in event_route[dialog:start] does not work
by Pranathi Venkatayogi 21 Apr '17

21 Apr '17

Hi, I wanted to make an http_async_query in event_route[dialog:start] after the call the confirmed. I see that flow does not even get to making "async_send_query". Perhaps it is failing at the first line "if (msg==null)" in w_http_async_post as I don't see any other error messages from the module. I can confirm that my setup and api server are fine as other calls I make from request_route and event_route[dialog:end] are working. I am attaching herewith my config files and Kamailio.log file. Relevant prob spot is below- Apr 20 17:37:17 devsip00 /sbin/kamailio[6286]: {2 3094 INVITE d9695149ce484811ab98ca41a4e8c694} INFO: <script>: Translation: TextBroker - Making CallAnswered for d9695149ce484811ab98ca41a4e8c694 If you notice we don't see any further progress of this call in http_async_client module. Can someone suggest what I am missing/where else can I know for sure dialog is confirmed and make this call. Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

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Kamailio 5.0 doesn't forward Authorization (Digest username) to Asterisk
by gh 21 Apr '17

21 Apr '17

Hi Marko, i work on the same scenario. It should work with: if(uac_reg_request_to("$fU", 0)) { xlog("L_NOTICE", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); t_relay(); } But in my case xlog() currently does not print out the variables, so i am not able to debug:-( Can you try this code and tell me if xlog() works for you? Best regards Georg

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Planning next IRC devel meeting - Apr 20, 2017
by Daniel-Constantin Mierla 20 Apr '17

20 Apr '17

Hello, probably it is the time to have another IRC devel meeting to discuss the plans for the project and the roadmap to next major release, v5.1. First proposed date would be Apr 20, with secondary options for Apr 24 or 25, but other dates can be taken in consideration if more convenient for a consistent group of developers. I created the wiki page for it at: * https://www.kamailio.org/wiki/devel/irc-meetings/2017a Feel free to add there the subjects you want to discuss, I also have several other topics to add. It would be good to list yourself if you are going to participate (or want to participate) in a specific day in order to see who are we relying on being present. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Reply to Incoming SIP Options Pings from Kamailio
by SIP Tech 20 Apr '17

20 Apr '17

Hello, I have a simple Kamailio server built that acts as a stateless load balancer for incoming SIP calls. All kamailio does here is receive the incoming call and route it round robin to a Freeswitch machine behind it. All configuration in Kamailio is done in local config files including the use of dispatcher.list. I have an external 3rd party partner sending my Kamailio machine SIP options pings. Because my Kamailio sends all incoming calls stateless to a FS machine behind it (round robin) the FS machines behind Kamailio get the incoming ping requests and generate the 200 OK reply back to kamailio. I would like the 3rd party to get a 200 OK for pings directly from my Kamailio and not from freeswitch behing it. I have the need to scale FS machines on the fly so I don't want FS machines to be responsible for the sip options ping replies. I want Kamailio itself to reply with a static 200 OK to incoming ping requests. I have looked into nathelper and siputils options_reply to see if I could get a simple static config to handle this 200 OK directly from Kamailio. I cant get it to fully work as the incoming sip pings still hit freeswitch. How can I configure this in the kamailio.cfg to get kamailio itself to reply with a 200 OK to SIP Options Pings and NOT forward the ping request to freeswitch? Here is my current simple stateless LB config in kamailio that works great for what it is and for what I need for loadbalancing incoming call requests. route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "4"); forward(); #stateless forward # t_relay(); #stateful relay } Here is a sample of what the incoming pings look like that hit Kamailio. U 2017/04/19 21:21:00.89 192.168.168.168:5060 -> 172.31.31.31:5060 OPTIONS sip:172.31.31.31:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.168.168:5060;branch=z9hG. Call-ID: 536f9b34c(a)192.168.168.168. To: sip:ping@172.31.31.31. From: <sip:ping@192.168.168.168>;tag=2b769f. Max-Forwards: 70. CSeq: 91160 OPTIONS. Route: <sip:172.31.31.31:5060;lr>. Using the above info what would a route look like in kamailio.cfg for this? What specific modules need to be installed/enabled to make your suggested route functional? Thanks guys

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