sr-users April 2019

sr-users@lists.kamailio.org

81 participants
110 discussions

Kamailio stop to process incoming SIP traffic via TCP.
by Kristijan Vrban 16 Jan '20

16 Jan '20

Hi kamailios, i have a creepy situation with v5.2.1 stable Kamilio. After a day or so, Kamailio stop to process incoming SIP traffic via TCP. The incoming TCP network packages get TCP-ACK from the OS (Debian 9, 4.18.0-15-generic-Linux) but Kamailio does not show any processing for the SIP-Traffic incoming via TCP. No logs, nothing. While traffic via UDP is working just totally fine. When i look via command "netstat -ntp" is see, that the Recv-Q get bigger and bigger. e.g.: Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 4566 0 172.17.217.12:5060 xxx.xxx.xxx.xxx:57252 ESTABLISHED 31347/kamailio After Kamailio restart, all is working fine again for a day. We have maybe 10-20 devices online via TCP and low call volume (1-2 call per minute). The only settings for tcp we have is "tcp_delayed_ack=no" How to could we debug this situation? Again, no error, no warings in the log. Just nothing. Kristijan

11 46

WebRTC ACK Protocol
by Ilie Soltanici 08 Jan '20

08 Jan '20

Hello, I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine. So far, everything is working fine, I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. This is how invite is looking: INVITE sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss SIP/2.0 Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060> Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE User-Agent: Proxy Date: Wed, 03 Apr 2019 17:11:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Alert-Info: Content-Type: application/sdp Content-Length: 596 Server: SIP Proxy and then WebRTC app is replying with 200 OK: SIP/2.0 200 OK Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE Contact: <sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Proxy-WEBRTC Content-Type: application/sdp Content-Length: 901 and finally, Kamailio is trying to send this ack through UDP protocol: ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0 Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport Route: <sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes> Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 ACK User-Agent: Proxy Content-Length: 0 If i'm trying to force it through TLS, i'm receiving error: get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443, to udp:22.22.22.22:23317) Can someone point me in the right direction, please? Thank you.

3 4

rtpproxy does not response, disable it
by Zodiac 03 Jan '20

03 Jan '20

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

5 10

Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

3 5

Best practises for websocket authentication
by Grant Bagdasarian 12 Jun '19

12 Jun '19

Hello, I was wondering if there are any best practices for authenticating websocket connection establishment inside the xhttp event route? Thanks and regards, Grant

3 3

forcing socket doesn't work for ACK
by David Villasmil 03 May '19

03 May '19

Hello guys, I have public and private IPs, and i need to force the sending socket replying to internal out externals endpoint, so i'm using this: function to check the destination (i tried using $sndto(ip) but it always returns NULL): route[CHECK_DEST_NET] { if ( $du=~"sip:172\..*" ) { setflag(FLAG_TO_PRIVATE); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PRIVATE -> [$du]\n" ); } else { setflag(FLAG_TO_PUBLIC); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PUBLIC -> [$du]\n" ); } return; } on my relay: route(CHECK_DEST_NET); if (isflagset(FLAG_TO_PRIVATE)) { xlog("L_ERR", "[RELAY] forcing socket to PRIVATE NET\n" ); force_send_socket(LISTEN_INSIDE_IF:LISTEN_INSIDE_PORT); } else { xlog("L_ERR", "[RELAY] forcing socket to PUBLIC NET\n" ); force_send_socket(LISTEN_OUTSIDE_IF:LISTEN_OUTSIDE_PORT); } This works nicely, except for ACKs and potentially other packets sent statelessly? Anyone knows how to fix this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

6 26

CSeq not incrementing - app_ruby
by Andrew White 01 May '19

01 May '19

Hi all, I’m still building with app_ruby and loving it - for the most part! I’m having an issue with one of my providers responding with a 482 on auth invite. Researching, it appears that my CSeq isn’t incrementing and this is the reason for the issue. I already have the relevant flag set in my kamailio.cfg however: modparam("dialog", "track_cseq_updates", 1) During my uac_auth, I’ve tried manually setting the diff, as well as running msg_iflag_reset (as suggested by Daniel - https://github.com/kamailio/kamailio/issues/1359 <https://github.com/kamailio/kamailio/issues/1359> - unfortunately the function isn’t exported in KEMI): if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") #KSR::COREX.msg_iflag_reset("UAC_AUTH") KSR::PV.sets("$dlg_var(cseq_diff)", "1") KSR.info("CSeq diff: #{KSR::PV.gete("$dlg_var(cseq_diff)")}") KSR::TM.t_relay() Unfortunately in both cases the actual CSeq doesn’t increment in the second INVITE: INVITE 1 (unauthenticated): > CSeq: 102 INVITE INVITE 2 (with Authorization header): > CSeq: 102 INVITE I’m unsure if I’ve missed something silly around the track_cseq_updates flag, or if app_ruby somehow isn’t interacting with the dialog module when running uac_auth to trigger the increment? Thanks! Kind regards, Andrew

3 6

listening at link local IPv6 address
by Juha Heinanen 01 May '19

01 May '19

My K host is behind a mobile WiFi hotspot and I tried to make K listen at the IPv6 address fe80::6e29:95ff:fe7d:37e6/64 that the host has received from the hotspot. Using that address on listen line gives error: Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:395]: udp_init(): bind(8, 0x7ff97dfb3fe4, 28) on FE80:0:0:0:6E29:95FF:FE7D:37E6: Invalid argument Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:401]: udp_init(): might be caused by using a link local address, try site local or global I can ping that address like this: $ ping6 fe80::6e29:95ff:fe7d:37e6%wlp1s0 PING fe80::6e29:95ff:fe7d:37e6%wlp1s0(fe80::6e29:95ff:fe7d:37e6%wlp1s0) 56 data bytes 64 bytes from fe80::6e29:95ff:fe7d:37e6%wlp1s0: icmp_seq=1 ttl=64 time=0.030 ms I then tried to add %wlp1s0 suffix to the listen address, but got error: Apr 21 09:10:35 salmon sip-proxy[7896]: 0(7948) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 408, column 34: ip address, interface name or hostname expected Did I do something wrong? If not, why is it that K cannot listen at a link local address? It is something that Linux networking stack does not support? -- Juha

6 33

Pipelimit with clean_unused
by Cinthia Leung 30 Apr '19

30 Apr '19

Hello, What is your recommendation on using clean_unused with db defined pipes? >>pl_check("systemwide"); If I leave the system idle, they are eventually "cleaned" as expected. In our use case, that means failed calls unfortunately. I am trying to see if I can mix both db/static pipes with dynamic/run-time pipes. Thanks Cindy

1 1

Next hop of reply, in onreply_route
by Alex Balashov 30 Apr '19

30 Apr '19

I am fairly sure it’s been asked a number of times, but is there a way to know the destination, and more especially the transport, of the next hop of a reply message, viewable from onreply_route? For requests, we have $nh and many other PVs which in some way evidence where it’s going. And yes, I’m aware that this can be previewed from the onsend_routes*, but I need to amalgamate that knowledge with some transaction-specific data. Thanks! — Sent from mobile, with due apologies for brevity and errors.

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sr-users April 2019