sr-users February 2018

sr-users@lists.kamailio.org

90 participants
109 discussions

How to add a new module to existing Kamailio Server?

by Arun NV

I am using a *Kamailio 4.4* server. And I would like to install *http_async_client* module to this server. I have looked in the module directory(*/usr/local/lib64/kamailio/modules*) to check whether this *http_async _client* module already exists there, but it was not there. And I searched for a solution and only found <http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-modular-install> the method to include new module at the compile time. I could find the modules source code on GitHub <https://github.com/sipwise/kamailio/tree/master/modules/http_async_client>. But I don't know the steps to install a new module to the existing Kamailio Thank you, Arunbalan

6 years, 2 months

Use RTPProxy in 302 scenario

by Fabian Borot

I have this scenario Customer --> Kamailio --> Provider1 The route where l route the call to Provider1 is called route["INCOMING"] and I "arm" the replies and failures to : t_on_reply("INCOMING") and t_on_failure("INCOMING"). Provider1 always replies with a 302 Redirect and in the failure_route["INCOMING'] I have this code: if (t_check_status("302")) { get_redirects("*"); t_on_branch("REDIRECT_BRANCH"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL_FORKING"); t_relay(); exit; } In the t_on_failure("SERIAL_FORKING") block I do t_relay() as long as there are more contacts to be processed. Lets say I need to try Provider_2 and Provider_3 failure_route["SERIAL_FORKING"] { xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Reply [$rs]\n"); if (!t_next_contacts()) { exit; } xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Trying next Contact.\n"); t_on_failure("SERIAL_FORKING"); t_relay(); } I want to use RTPProxy in this scenario and I am using the function rtpproxy_manage() to let kamailio handle it automatically. RTPProxy ↑ | | Customer --> Kamailio --> Provider1 | | ↓ Provider N After the 302 response is processed I would like to have the signaling flow "Customer -> Kamailio --> ProviderN" and the RTP flow "Customer --> RTPProxy --> ProviderN" I have tried inserting the function in several places but can't make it work and what I mean is that sometimes the rtpproxy creates the 1st session and then deletes it as soon as I call it the 2nd time with a message similar to this one (log of rtpproxy) : "INFO:416f20b74a58fb75168175e61b2edece@192.168.1.61:5060:rtpp_cmd_delete_ematch: forcefully deleting session 1 on ports 63502/0". Sometimes the kamailio log shows this message: "ERROR: rtpproxy [rtpproxy.c:2536]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" I think I must call rtpproxy_manage() before I send the INVITE to each of the Providers in the Contact list so they receive a SDP offer with an IP:Port combination provided by the RTPProxy (instead of the ones coming from the Customer). That would be in my route["INCOMING'] Then once Provider_N replies with a 183 with SDP or a 200 OK I would call it again to get a new session to be passed on the reply to my customer. But is the part I am struggling with. Should I add a new reply route inside the failure_route["INCOMING'] only if I am dealing with 302 and do it there ? I am out of ideas guys and any idea is welcome

6 years, 2 months

Can rtpengine change network interface of estabished call?

by Yufei Tao

Hi, Is it possible to define two pairs of network interfaces for rtpengine, e.g. int/ext and up/down? If so during a call established between interfaces 'int' and 'ext', is it possible to send a re-INVITE from the calling party on interface 'ext', to update the RTP destination to an IP on a different interface 'up', for example? Or does the updated RTP destination have to be on the same interface as when the call was established? I have some success updating the RTP destination if it is on the same interface (different IP:port) however haven't managed to divert it to a different interface. I wonder if it is possible at all or me missing something? Cheers, Yufei

6 years, 2 months

(no subject)

by Tihomir Culjaga

Hi, Im new to Kamailio so im sorry if some of my questions will sound a bit wierd :=) i have a setup like this: (Kamailio version 5.1) UA > Kamailio > Freeswitch Kamailio forwards registration to freeswitch without saving location. I have configured sip over websockets and im able to register to Freeswitcha via kamailio. Also, i use add_path_received() so kamailio adds a path header to register messages recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809: ------------------------------------------------------------------------ REGISTER sip:192.168.50.60 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 Max-Forwards: 69 To: <sip:1001@192.168.50.60> From: <sip:1001@192.168.50.60>;tag=vq30modpgd Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Authorization: Digest algorithm=MD5, username="1001", realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90", uri="sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef", qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001 Contact: <sip:1001@192.0.2.110 ;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, 100rel, outbound User-Agent: SIP.js/0.7.0 BB Content-Length: 0 Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6> ------------------------------------------------------------------------ send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 From: <sip:1001@192.168.50.60>;tag=vq30modpgd To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600 Date: Fri, 09 Feb 2018 21:38:07 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Content-Length: 0 ------------------------------------------------------------------------ Once, registered, im trying to originate a call to the registered endpoint from freeswitch this results in an invite like this: send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549: ------------------------------------------------------------------------ INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0 Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Max-Forwards: 70 From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK To: <sip:1001@192.0.2.110;transport=ws> Call-ID: 8212b700-8885-1236-08b3-000c293ae94e CSeq: 118770326 INVITE Contact: <sip:mod_sofia@192.168.50.65:15060> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit 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: 891 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:1002@192.168.50.65 >;party=calling;screen=yes;privacy=off Freeswitch appends a route header to the invite specifying kamailio as route and a received with original uri. well, here is the issue ... im failing to hit the SIP over websocket client... Kamailio my RELAY route is like this route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_ALERT", "Its a Branch, go manage that !!! <$ru>\n"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_ALERT", "its onreply_route, go manage !!! <$ru>\n"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_ALERT", "Its failure_route, go manage !!! <$ru>\n"); t_on_failure("MANAGE_FAILURE"); } } $fs = "udp:192.168.50.60:5060"; xlog("L_ALERT", "Trying to rely !!! <$ru>\n"); if (!t_relay()) { xlog("L_ALERT", "failed to rely !!! <$ru>\n"); sl_reply_error(); } exit; } here is a part of the log: Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:192.168.50.60:5060, to ws:192.0.2.110:5060) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core> [core/msg_translator.c:424]: clen_builder(): content-length: 916 (916) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core> [core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not create Via header Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26 n=xlog Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT: <script>: failed to rely !!! <sip:1001@192.0.2.110;transport=ws> is that because i set a $fu before t_relay() ? any idea how to approach the issue ?

6 years, 2 months

onend_route() ?

by Wilkins, Steve

I meant to reply to Kamailio mailing list. From: Wilkins, Steve Sent: Friday, February 9, 2018 12:40 PM To: 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: [SR-Users] t_relay()? Hi Daniel, I am not sure what I am doing wrong but I cannot get the forwarded message to print. I am using onend_route() and trying to print the results using xdbg. I have searched for examples and have had no luck. Thank you From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Friday, February 9, 2018 7:48 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Wilkins, Steve <swwilkins(a)mitre.org<mailto:swwilkins@mitre.org>> Subject: Re: [SR-Users] t_relay()? Hello, before t_relay() you can't see what is going to the network, because some headers are added when knowing target address and local socket (e.g, Via, Record-Route). You can use onsend_route for seeing the request just before writing to the network. https://www.kamailio.org/wiki/cookbooks/devel/core#onsend_route $snd(buf) will give outgoing buffer -- for more see $sndfrom(...) and $sndto(...) vars. Cheers, Daniel On 09.02.18 11:24, Wilkins, Steve wrote: Hi All, Could someone please tell me, what variable to I use to see what t_relay() fowarding in its message? Thank you, -Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com<http://www.asipto.com> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com<http://www.kamailioworld.com>

6 years, 2 months

Request Timeouts

by Wilkins, Steve

Hello, This is actually a follow up to my last question about some providers calls being dropped. I noticed that when calls are dropped, I see a Kamailio log message "Request Timeout" about 5-10 seconds before the call is dropped. I am wondering what could cause this while a call is in progress. Does Kamailio need to respond to Asterisk with a '200 OK' for an INFO request sent to Kamailio from Asterisk? I am not sure what else could cause this call dropping behavior. Note: It does not appear to be caused from a missing 'ACK'. Thank you, -Steve

6 years, 2 months

What is the mechanism used for sending push notification from Kamailio, For both IOS and Android?

by Arun NV

I am creating custom phone client, which uses *Kamailio *as a proxy and it's required to send *push notification* from *Kamailio *to the clients when a call towards the client reached on *Kamailio*. I found a module for APNS here* https://github.com/tvntsr/push <https://github.com/tvntsr/push>*. But I am developing both Android and IOS clients. So What mechanism I can use for the Android side? Is there any module available for *GCM*(Google Cloud Messaging) in Kamailio? Thank you, Arunbalan

6 years, 2 months

is_audio_on_hold() doesn't support rfc3264

by Julia Boudniatsky

FYI, After some tests found, that function "*is_audio_on_hold()*" (textops) *supports only rfc 254*3, where "hold" is indicated by setting the "c" destination addresses for the media streams to zero (0.0.0.0). The newest *rfc3264*, where "hold" is indicated by a=sendonly/inactive, *is not supported*. Kamailio version 4.4.6. Best regards, Julia

6 years, 2 months

Load testing: Taking CPU & MEM stats

by jeevan Chaitanya

Hi All, Found python library to take system stats during the load. >>> from getstats import GetStats >>> stat_obj = GetStats.getstats(dur=300,rate=10) >>> stat_obj.run_stats() Running.......... ***** Completed capturing the STATS ***** ######################################## # stats_2018-02-12_16-47-47_.csv # ######################################## Doc URL: http://getstats.readthedocs.io/en/latest/ If you require use and let me know. Thanks, K.Jeevan.

6 years, 2 months

Very inconsistent calls

by Wilkins, Steve

Hello, I have found that calls are very inconsistent. I use Kamailio 5, Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls into a WebRTC client (tryit-jssip), sometime the calls stay up until I close them (10-15 minutes), others times those calls drop in 30 seconds. This is extremely confusing...does anyone else experience this type of behavior? -Steve

6 years, 2 months

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sr-users February 2018