sr-users June 2015

sr-users@lists.kamailio.org

82 participants
136 discussions

How to disable retransmits via TCP connection? etc.
by Andrey Utkin 25 Jun '15

25 Jun '15

Hi! We meet such an issue with some kinds of client internet connections (e.g. reproduces always on local wifi, but doesn't repr. on 3g inet). We don't know what are exact network characteristics, but what happens from perspective of traffic on SIP server is the following: 1. the SIP client registers successfully to Kamailio (which is recent git master HEAD) with TCP transport for SIP; 2. another client calls this user; 3. Kamailio relays INVITE, it gets transmitted fully and ACK-ed on TCP level; 4. then, within 0.1-0.2 second, Kamailio fires first retransmit of that INVITE, which doesn't get fully transmitted (it is shown trunkated in sniffer output, no TCP ACK replies for it); 5. Kamailio tries to retransmit INVITE again, with the same result as in #5. 6. The caller gets SIP/2.0 408 Request Timeout. A piece of traffic from "ngrep -t -e -d any -W byline port 5060": https://gist.githubusercontent.com/krieger-od/219f9975e5efb980ff5b/raw/096a… The called side app is based on mobile Linphone app. Switching to UDP is not an option (in some networks SIP messages get delivered trunkated which breaks calls), we will check how it works with TLS transport a bit later when there's technical possibility. I have two questions: 1. How would I completely disable retransmissions to TCP connections? 2. Any ideas what can be the reason for this issue? Retransmissions by themselves? -- Andrey Utkin

4 11

Kamailio Server Maintenance - Wed-Thu Night
by Daniel-Constantin Mierla 25 Jun '15

25 Jun '15

Hello, during the night between Wed (June 24) and Thu (June 25), planned to start not early than 00:00 GMT+1, there will be some schedule maintenance work to the infrastructure that is hosting some of the kamailio.org servers. The main affected services will be: - main website (www.kamailio.org) - wiki portals - mailing lists (lists.sip-router.org) - git mirror (git.kamailio.org) It is expected to have short downtimes of few minutes. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Kamailio + RTPEngine + Asterisk
by Javier Aristizábal 24 Jun '15

24 Jun '15

Hello folks, I am trying to configure this on the same server: Kamailio v4.2 port 5060 RTPEngine v2.3 port 22222 Asterisk v11.18.0 port 5080 I am follow the Asipto tutorials, for the Kamailio and the real time configuration. Now, I can talk between extension and PSTN without issues. But the trouble is with the Status registration with Asterisk. And I need that Status OK because I am using queues in asterisk for call centre. I got this from with: sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 101 (Unspecified) D Yes Yes 0 UNREACHABLE Cached RT the status is UNREACHABLE and the asterisk table sipregs won't update after each registration. To be sure, I tried first all config only with asterisk to check that is not a problem with MySQL or something. On my cfg file I have this for the asterisk registration: route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } Any clue? Thanks! -- Javier Aristizábal

2 16

Kamailio proxy at Amazon EC2 (behind NAT)
by Alexandru Covalschi 24 Jun '15

24 Jun '15

Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP. In case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 15

Kannot start kamailio 4.3
by Yuriy Gorlichenko 23 Jun '15

23 Jun '15

Try to start kamailio on Ubuntu 14.04.02 Get this errors ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) my params for ctl # ----- ctl params ----- modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 1dbd53 compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4

4 9

kamailio fails after start
by Yuriy Gorlichenko 22 Jun '15

22 Jun '15

Hello. I Installed kamailio on ubuntu 14.04 that runs as virtual systemOpenVZ. after starting kamailio I see that it runs ok with kamailio start or kamctl start at ps -ax I see working processes. But after one minute of working kamailio fails with kamailio: ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to pipe I see nothing more at debug file. No errors. (I run with debug=4) my cfg parameters at start mhomed=1 memdbg=5 memlog=5 fork=yes children=6 version: kamailio 4.2.4 (x86_64/linux) ac3682 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: ac3682 compiled on 04:27:04 Apr 10 2015 with gcc 4.8.2

2 2

Kamailio transformation
by Ali Taher 22 Jun '15

22 Jun '15

Hello, Suppose I have $var(a) = 1234#5678 and $var(b)=123455667 I need to get the number before '#' character if exists , meaning that from $var(a) I must get 1234# and from $var(b) I must get empty string. I tried $var(pref)= $(var(a){s.select,0,#}); but I'm getting the whole string if # doesn't exists. There is a way to check if # exists by subtracting the length of the string before and after removing # character and check if length changes , but I want a cost-less method. Any help would be appreciated. Thanks Ali

3 2

Capture a Match from Regex comparison
by Sebastian Damm 22 Jun '15

22 Jun '15

Hi, we have some cases, where we get multiple PCMA lines with different numbers from a peer. One of our backend systems doesn't handle that correctly. So what I want to do is detect whether there is a PCMA line with a non-standard number in the SDP. This is how an SDP looks like: v=0 o=hiQ9200 2990220150519102450 1229717573 IN IP4 1.2.3.4 s=Phone Call c=IN IP4 1.2.3.4 t=0 0 m=audio 40174 RTP/AVP 8 0 18 96 97 13 100 98 99 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:0 PCMU/8000 a=fmtp:0 vad=no a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 G726-32/8000 a=rtpmap:97 AAL2-G726-32/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=rtpmap:98 PCMA/8000 a=gpmd:98 vbd=yes a=rtpmap:99 PCMU/8000 a=gpmd:99 vbd=yes a=sqn: 0 a=cdsc: 1 image udptl t38 a=sendrecv a=pmft: T38 a=ptime:20 I tried something like that (expecting that the number doesn't change): if (sdp_get_line_startswith("$avp(badcodec)", "a=rtpmap:98 PCMA")) { But this makes Kamailio drop the INVITE when arriving at the line in the config. I guess, "startswith" stops parsing the line at the colon. So I thought I'll make it more generically: sdp_get("$avp(sdp)"); if ($avp(sdp) =~ "a=rtpmap:([1-9][0-9][0-9]?) PCMA") { sdp_remove_codecs_by_id("$missingvariable"); } What I need is to capture the number in the a=rtpmap line so I can delete the codec by ID. But I couldn't find a way to capture the matches from my regex comparison. Am I missing something? (Other approaches for my problem would be welcome, too.) Thanks in advance. Best Regards, Sebastian

2 3

Kamailio with Blox SBC (opensips based) - Wrong CSeq
by kai.ohnacker＠cbc.de 22 Jun '15

22 Jun '15

Hello, my setup is a kamailio server as registrar and Blox as SBC. Blox is a freeware GUI with Opensips as framework. Kamailio is located in the private network and Blox operates with 2 NICs (private and public) Now the following problem: If I make a call from public (phonerlite) to private (mayah), the connection don’t reach the “framed” state and is “broken” after 30 sec. After evaluating the log files, I found that the CSeq from Kamailio is wrong. The invite comes from CSeq: 9 INVITE The reply is with CSeq 1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.3:6060;branch=z9hG4bK38ae.cf317344.0 Max-Forwards: 70 To: <sip:1001@192.168.1.3:8000>;tag=qTKGZvr0ItW From: "PhonerLite" <sip:1000@192.168.1.3:8000>;tag=1005520878 Call-ID: SBCbFgsQn5HcHpEUxJUVF8gQWVZREt6QwJvWVJkXCdAUzdkXl41CUJ4cEdWZV1NQ0tFYQ CSeq: 1 Contact: <sip:1001@192.168.0.33:5060> Record-Route: <sip:192.168.0.1;lr=on> User-Agent: MAYAH 4.9.12.0-2.1.0.45 Content-Length: 0 And here the information from the opensips log files: Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq: no method found Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq: bad cseq Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:get_hdr_field: bad cseq Perhaps someone of you knows this failure or can tell me how to solve this problem? Cheers, Kai

3 4

Redirect with path
by Kelvin Chua 22 Jun '15

22 Jun '15

on my registrar server: after calling lookup, and getting AOR with path, how do I send the path info back as a redirect? Kelvin Chua

2 2

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sr-users June 2015