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37519 discussions

Confusion about various TCP expiry and lifetime parameters
by M S 27 Aug '15

27 Aug '15

Hi, We have a bit of confusion regarding tcp_connecton_timeout core parameter. The documentation says, -- Lifetime in seconds for TCP sessions. TCP sessions which are inactive for longer than *tcp_connection_lifetime* will be closed by Kamailio. -- However we observe a strange behaviour. 1. The connection is NOT closed by Kamailio unless we additionally set "close_expired_tcp" parameter in usrloc module, 2. Secondly, if we set this parameter to a smaller value say 70 seconds while sip register expiry is 600 seconds (and close_expired_tcp parameter enabled in usrloc module), the connection still remains active (tested after keeping it idle for 120 seconds then sending a sip request on this connection, we expected the request to fail but it does not fails). 3. if we set this parameter to greater then sip register value, e.g. 610 seconds and set close_expired_tcp parameter in usrloc, then we observe disconnect after about 8-10 minutes. Whereas we expect it to continue since user is re-registering every 600 seconds. Can you guys explain why this is happening? What keeps a tcp connection active or makes it inactive? We are using websockets (which use TCP at lower layer) and we observe there is no websocket frame sent or received over the tcp connection, yet it remain active after tcp_connection_lifetime in case 2 above and becomes inactive in case 3. We are using Kamailio v4.3.1 SVN Rev. 4717b5 on Debian Wheezy 32bit OS. Please suggest. Thank you.

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IMS Kamailio "477 Unfortunately error on sending to next hop occurred" code sent by the PCSCF
by Victor Huertas Garcia 26 Aug '15

26 Aug '15

Hello everybody, I have tried to perform a call between alice and bob (both registered in the same SCSCF and from the same PCSCF). The INVITE goes all the way as it is supposed: Alice-->PCSCF-->SBC-->SCSCF-->ICSCF-->SCSCF-->SBC-->PCSCF. But when it reaches PCSCF on Network to User direction, this SIP proxy answers with a "477 Unfortunately error on sending to next hop occurred" code just after an exchange with the RTP Proxy (Offer-Answer). At kamailio.cfg (which it is not change from the default config) this exchange is done under Route[RTPRPROXY] in the function: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove"); Does anybody know why is complaining (see below) about the server on localhost while it is not configured so? Here you can find the logs from PCSCF: Aug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Received command 'offer' from 192.168.0.108:57522Aug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Creating new callAug 23 15:10:06 ims rtpengine[556]: [3D504184-55D9D43E00072478-7E0DE700] Replying to 'offer' from 192.168.0.108:57522Aug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: *** cfgtrace:request_route=[RTPPROXY] c=[/etc/kamailio/pcscf/kamailio.cfg] l=813 a=2 n=returnAug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: *** cfgtrace:request_route=[Term_Initial] c=[/etc/kamailio/pcscf/kamailio.cfg] l=1196 a=24 n=t_relayAug 23 15:10:06 ims /usr/sbin/kamailio[3313]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f480e27ed70,1358,0,192.168.0.105:49284,16): Invalid argument(22)Aug 23 15:10:06 ims /usr/sbin/kamailio[3313]: CRITICAL: <core> [udp_server.c:555]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Here you can find the configuration from RTP proxy (/etc/default/ngcp-rtpengine-daemon) and you can see that I am not listening the localhost port at "INTERFACES": RUN_RTPENGINE=yes LISTEN_TCP=25060LISTEN_UDP=12222LISTEN_NG=22222LISTEN_CLI=9900INTERFACES="192.168.0.108"TIMEOUT=60SILENT_TIMEOUT=3600PIDFILE=/var/run/ngcp-rtpengine-daemon.pidFORK=yes# TOS=184TABLE=0# NO_FALLBACK=yes# PORT_MIN=30000# PORT_MAX=50000# REDIS=127.0.0.1:6379# REDIS_DB=1# B2B_URL=http://127.0.0.1:8090/# LOG_LEVEL=6# LOG_FACILITY=daemon# LOG_FACILITY_CDR=daemon# LOG_FACILITY_RTCP=daemon# NUM_THREADS=5# DELETE_DELAY=30# GRAPHITE=9006# GRAPHITE_INTERVAL=60# GRAPHITE_PREFIX=myownprefix# MAX_SESSIONS=5000 And finally the pcscf.cfg: # SIP / UDPlisten=udp:eth0:4060#listen=udp:127.0.0.1:4070listen=udp:127.0.0.1:4060# SIP / TCP (Monitoring)#listen=tcp:127.0.0.1:4060# SIP / TCP/TLS#listen=tls:109.239.57.200:5061# SIP / Websocket #!define MY_WS_PORT 80 listen=tcp:eth0:MY_WS_PORT # SIP / Websocket/TLS#!define MY_WSS_PORT 443#listen=tls:109.239.57.200:MY_WSS_PORT #alias=pcscf-1.imscore.orgalias=pcscf.home-domain.net#alias=proxy.imscore.org#alias=tls:"wss-proxy.imscore.org":443 # Port, where we listen to Traffic#!define PORT 4060 # NUEVO Workshop!#!define SBCPORT 5080#!define PCSCF_URL "sip:pcscf.home-domain.net"#!define TCP_PROCESSES 32 #!subst "/NETWORKNAME/home-domain.net/"#!subst "/HOSTNAME/pcscf.home-domain.net/"#!define HOSTNAME_IP pcscf.home-domain.net#!define HOSTNAME_ESC "pcscf\.home-domain\.net" # SIP-Address of capturing node, if not set, capturing is disabled.##!define CAPTURE_NODE "sip:10.1.8.55" # Allowed IPs for XML-RPC-Queries#!define XMLRPC_WHITELIST_1 "127.0.0.1"##!define XMLRPC_WHITELIST_2 "127.0.0.1"##!define XMLRPC_WHITELIST_3 "127.0.0.1" # Databases:#!define DB_URL "mysql://pcscf:heslo@127.0.0.1/pcscf"##!define DB_URL2 "con2=>mysql://pcscf:heslo@127.0.0.1/pcscf"##!define DB_URL "con1=>mysql://pcscf:heslo@192.168.5.1/pcscf"##!define DB_URL2 "con2=>mysql://pcscf:heslo@10.1.27.217/pcscf" #! Optional: Server-URL for Websocket-Requests##!define WEBSOCKET_WEBSERVER "phone.imscore.org" # NUEVO Workshop!#!subst "/ICID_VALUE_PREFIX/P-CSCFabcd/"#!subst "/ICID_GEN_ADDR/127.0.0.1/" # IP-Adress(es) of the RTP-Proxy##!define RTPPROXY_ADDRESS "udp:10.1.2.186:22222 udp:10.1.27.217:22222"#!define RTPPROXY_ADDRESS "udp:192.168.0.108:22222"## Several features can be enabled using '#!define WITH_FEATURE' directives:## *** To run in debug mode:# - define WITH_DEBUG## *** To enable nat traversal execute:# - define WITH_NAT# - define the connection to the RTP-Proxy: RTPPROXY_ADDRESS## *** To force alls calls through the RTP-Proxy# - this will automagically enable NAT-Traversal# - define FORCE_RTPRELAY## *** To enable IPv4/IPv6 Translation (RTPProxy)# - this will automagically enable NAT-Traversal# - define WITH_RTPIPV4## *** To enable TCP support execute:# - define WITH_TCP## *** To enable TLS support execute:# - adjust CFGDIR/tls.cfg as needed# - define WITH_TLS# - this will automagically enable TCP## *** To enable XMLRPC support execute:# - define WITH_XMLRPC# - this will automagically enable TCP## *** To enable anti-flood detection execute:# - adjust pike and htable=>ipban settings as needed (default is# block if more than 16 requests in 2 seconds and ban for 300 seconds)# - define WITH_ANTIFLOOD## *** To enable the Rx-Interface:# - Configure Rx-Diameter-Interface in pcscf.xml# - define WITH_RX## *** To enable a Homer SIP-Capter-Node:# - define CAPTURE_NODE with a proper address## *** To enable support for the SEMS-SBC# - define WITH_SBC# - configure dispatcher-list with a set of SBC's## *** To enable support for Websocket# - define WITH_WEBSOCKET# - this will automagically enable TCP## Enabled Features for this host:#!define WITH_DEBUG#!define WITH_NAT#!define WITH_NATPING#!define FORCE_RTPRELAY##!define WITH_TLS#!define WITH_XMLRPC#!define WITH_ANTIFLOOD##!define WITH_RX##!define WITH_RX_REG##!define WITH_RX_CALL##!define WITH_TCP##!define WITH_SBC##!define WITH_RTPIPV4#!define WITH_SBC_CALL#!define WITH_REGINFO#!define WITH_WEBSOCKET#!define WITH_IMS_HDR_CACHE Thanks a lot for your attention

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Kamailio dead after INVITE from SIPDROID
by Kowalski Piotr 3 - Hurt 26 Aug '15

26 Aug '15

Hello Community. We have openIMS nodes in our lab. We have Kamailio 4.0 based solution. I registered SIPDROID client (iptel) on Android and JITSI on WINDOWS. When I call from SIPDROID to JITSI following happens: SIP DROID - iptel - client - registration ok, but INVITE attempt towards JITSI (and other clients) does what's below: HSS (every 30 seconds): [root@hss1 kowalp00]# 2015-08-25 12:16:55,468 WARN de.fhg.fokus.diameter.DiameterPeer.transport.Communicator - run Read 3021 before Exception. 2015-08-25 12:16:56,980 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf1.hss.ims.test.pl:3868 not responding to connection attempt 2015-08-25 12:17:26,985 ERROR de.fhg.fokus.diameter.DiameterPeer.peer.StateMachine - I_Snd_Conn_Req StateMachine: Peer scscf1.hss.ims.test.pl:3868 not responding to connection attempt SCSCF: Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8879]: INFO: <core> [main.c:854]: INFO: signal 15 received Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8865]: INFO: <core> [main.c:854]: INFO: signal 15 received Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: INFO: snmpstats [snmpstats.c:410]: The SNMPStats module got the kill signal Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: INFO: snmpstats [snmpstats.c:414]: Shutting down the AgentX Sub-Agent! Aug 25 12:24:18 scscf1 /usr/sbin/kamailio[8803]: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Aug 25 12:25:18 scscf1 /usr/sbin/kamailio[8803]: : <core> [main.c:697]: BUG: shutdown timeout triggered, dying... Kamailio becomes dead and needs restart. Clients become unregistered. A call from JITSI to SIPDROID it's fine. Any ideas? Piotr Kowalski Starszy Architekt Techniczny Tel. +48 22 699 50 93 Kom. +48 519 123 289 Orange Orange Polska Hurt Badania i Rozwój Sieci i Platform Usługowych Wydział Usług Konwergentnych Obrzeżna 7 /p. 309, 02-691 Warszawa Fax. +48 22 857 99 86 <http://www.hurt-orange.pl/> www.hurt-orange.pl - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Czy musisz drukować tę wiadomość? Pomyśl o środowisku. Korzystaj z e-faktury. Treść tej wiadomości jest własnością Orange Polska i zawiera informacje stanowiące tajemnicę przedsiębiorstwa Orange Polska. Jeżeli nie jesteście Państwo jej adresatem, bądź otrzymaliście ją przez pomyłkę, prosimy o powiadomienie o tym nadawcy oraz trwałe jej usunięcie. Orange Polska Spółka Akcyjna z siedzibą i adresem w Warszawie (02-326) przy Al. Jerozolimskich 160, wpisana do Rejestru Przedsiębiorców prowadzonego przez Sąd Rejonowy dla m.st. Warszawy XII Wydział Gospodarczy Krajowego Rejestru Sądowego pod numerem 0000010681; REGON 012100784, NIP 526-02-50-995; z pokrytym w całości kapitałem zakładowym wynoszącym 3.937.072.437 złotych.

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Rewrite Invite w/o sdp
by Serge Yuriev 25 Aug '15

25 Aug '15

Hello, Our PBX (Cisco CallManager) uses "late offer" sending invites w/o sdp. After receiving provisional reply 18x it answers prack with sdp I'm trying to realize following scenario: 1 Pbx (invite no sdp) 2 kamailio hold and saves invite 3 send 183 to Pbx 4 get prack with sdp 5 construct new invite from saved one and sdp from prack 6 send it to calee Is this possible? How to save invite for future use? -- Serge via mobile

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dispatcher gateway probing in 4.3
by Joseph Dickson 24 Aug '15

24 Aug '15

Greetings.. I'm fairly new to Kamailio, and am having fun so far... I'm using the latest 4.3 rpm version on Centos 7. I'm using Kamailio to front end a pair of FreeSwitch SBC boxes in an active/active config.. My plan is to use dispatcher to load balance calls between them. This is working fine.. the problem that I'm bumping into is that the gateway probing options in dispatcher seem not to fit my use case.. In addition to the two SBC gateways, I also have a few external gateways that I want to load balance between also using dispatcher. The problem is that I can't send options packets to these gateways -- they don't respond to them. This means I can't just configure dispatcher to probe all gateways (probing_method=1) because it will mark these gateways as inactive.. It looks like every other configuration of probing_method only probes a gateway until the state is determined, and then it stops probing. probing_method=2 does not appear to be used at all in the code.. Am I doing this wrong? Or is this just an area of dispatcher that needs some improvement? Looking at the code, I can probably add the functionality that I need, but I don't want to go that route if I'm just missing how this is usually done.. Thanks, -- *Joseph Dickson* E: jdickson(a)evolvetsi.com

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Request timout - Change value
by Loic Chabert 24 Aug '15

24 Aug '15

Hello everybody, I'm using kamailio as a dispatcher in front of asterisk boxes and i use a failure route if asterisk box does not respond or send 500error. failure_route[RTF_DISPATCH]{ if(t_is_canceled()){ exit(); } if(t_check_status("500") || (t_branch_timeout() && !t_branch_replied())){ xlog("L_WARN","[$fU@$si:$sp]{$rm} Asterisk Box $du is down\n"); ds_mark_dst("ip"); if(ds_next_dst()){ xlog("L_WARN","[$fU@$si:$sp]{$rm} Sending to Asterisk Box - $du\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } As you can see, i check if "t_branch_timeout() && !t_branch_replied()", but where has been stored timeout value ? How can i change it ? For exemple, i need to set 1s, and then, request expired. Thanks, Regards.

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Dialog module - db_fetch_rows
by jay binks 24 Aug '15

24 Aug '15

Hey All, Im just after someone else thoughts on if this is a bug. After reading : http://kamailio.org/docs/modules/3.3.x/modules_k/dialog.html#idp148408 Im lead to understand that setting *modparam("dialog", "db_fetch_rows", 0)* should instruct the dialog module to write dialogs to the DB, but not bother loading them on startup. This sounds like what I'm after. However ever time I set the value to 0, dialogs are still loaded from the DB. a quick dig in the source takes me to the function select_entire_dialog_table in dlg_db_handler.c this chunk of code does not appear to properly implement " if fetch_num_rows == 0 then do nothing " Im after a couple of things from you guys. a) validation that my understanding of what SHOULD happen is correct b) possibly validating that I'm not crazy, and this is actually a bug :) -- Sincerely Jay

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Ims Kamailio issue - "403 Forbidden - You must register first with a S-CSCF"
by Victor Huertas Garcia 23 Aug '15

23 Aug '15

Hello everybody, I have installed an IMS core using Kamailio and FHoSS. In this environment, I am able to successfully register Bob and Alice with the same S-CSCF. However, when I try to make a voice call between Bob and Alice it is not established. The code received is "403 Forbidden - You must register first with a S-CSCF". This code is generated by the P-CSCF because the Contact Header that is checked against the database ("location" table) it is not found. When I look up manually the "location" MySQL table, there are two things that surprise me: - The Contact header (AOR column content) that is included in the INVITE Requet URI when entering in the P-CSCF is present in the table. So why the pcscf_is_registered ("location") returns false? - When Bob registers two Contacts are stored, the first one is the one included in the REGISTER message generated from Bob IMS Client. However, the second Contact is the one present in the 200 OK message (coming from the S-CSCF) which is different from the one present in the REGISTER (the only thing different is the UDP port). Why the S-CSCF (with the default kamailio.cfg) doesn't the Contact header and changes the UDP port? Is this the standard behaviour? I don't think so... Thanks for your attention Regards,

2 3

A question to you, SIP providers: what is your policy on PBXes calling themselves
by Antonio Gómez Soto 23 Aug '15

23 Aug '15

Hello, This is question on PBX behavior, what is the right thing to do, and how do PBX's generally behave. If a user on a phone, dials a number, which happens to be configured on the same phone system (for example another tenant), there are two options: 1. The PBX notices this, and directly connects the phone to the DID on that system (breaking separation of tenants) 2. The PBX sends the call out on the SIP trunk, and the provider-routing sends the call back as an incoming call. What are the pros and cons of each option from the SIP provider point of view? How do PBX's generally behave? Thanks, Antonio PS: I reposted, because the original question was apparently not phrased in a clear way.

3 2

IPSec supporting open source SIP Server
by Priyaranjan Nayak 23 Aug '15

23 Aug '15

Hi All, I need one IPSec supported opensource SIP server for testing my sip client endpoint.Could you please tell me which SIP server supporting IPSec ? Thanks in advance. Thanks Priyaranjan

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