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37517 discussions

Cannot hear voice with symmetric NAT and STUN
by Khoa Pham 08 Feb '17

08 Feb '17

Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP. I read that STUN is for non-symmetric NAT, and RTP server is for symmetric NAT. Supposed A calls B. If A, B both use symmetric NAT and STUN, they cannot hear each other If A or B use non-symmetric NAT and NOT using STUN, they cannot hear each other. Why is that? I read http://tools.ietf.org/id/draft-takeda-symmetric-nat-traversal-00.txt for Prediction Failure, is that related to this problem ? -- Khoa Pham HCMC University of Science Faculty of Information Technology

8 18

Diversion Header - relay messed up
by Roman Dissauer 08 Feb '17

08 Feb '17

When I get an INVITE with Diversion Header the Request is forwarded without Diversion Header and the Request User is taken from Diversion User. Problem is that on the Destination Host I cannot get original Request User what is the intended destination! Is this intended behaviour? How can I change this behaviour? I already tried to delete the Diversion Header on request_route but this didn’t change the behaviour. Thanks in andvance for your help! Roman Here The INVITEs with x.x.x.x as Kamailio external IP and y.y.y.y as Carrier IP Incoming INVITE: INVITE sip:+43123456789@x.x.x.x:5060 SIP/2.0. Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 19. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Diversion: <sip:+436761234567@y.y.y.y:5060>;privacy=full;screen=no; reason=unconditional; counter=1. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 260. Content-Disposition: session; handling=required. Content-Type: application/sdp. ... Relayed INVITE: INVITE sip:436761234567@x.x.x.x:5060 SIP/2.0. Record-Route: <sip:10.23.101.1;r2=on;lr=on;ftag=gK022ac5cb>. Record-Route: <sip:x.x.x.x;r2=on;lr=on;ftag=gK022ac5cb>. Via: SIP/2.0/UDP 10.23.101.1;branch=z9hG4bK8582.4fc0216dbecafde29127db502993222c.0. Via: SIP/2.0/UDP y.y.y.y:5060;rport=5060;branch=z9hG4bK02B7871714d9345f843. From: <sip:+43987654321@y.y.y.y>;tag=gK022ac5cb. To: <sip:436761234567@x.x.x.x>. Call-ID: 906143813_44460603(a)y.y.y.y. CSeq: 28100 INVITE. Max-Forwards: 18. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS. Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed. Contact: <sip:+43987654321@y.y.y.y:5060>. P-Asserted-Identity: <sip:+43987654321@y.y.y.y:5060>. Supported: timer,100rel. Session-Expires: 1800. Min-SE: 90. Content-Length: 274. Content-Disposition: session; handling=required. ...

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Second coredump in a month
by Jon Bonilla (Manwe) 08 Feb '17

08 Feb '17

Hi all A server running version 4.4.4+0~20161223011227 has segfaulted for the second time this month. I'm attaching a bt in case someone can explain it to me. cheers, Jon

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kamailio with cassandra
by Никитенко Виталий 08 Feb '17

08 Feb '17

Hi! I want start kamailio with db_cassandra. do as it written in https://www.kamailio.org/docs/modules/4.2.x/modules/db_cassandra.html. Then trying kamctl start, failed with error "table version does not exist!". >From CQLSH table is view cqlsh:kamailio> Describe tables; version location cqlsh:kamailio> select * from version; table_name | table_version ------------+--------------- version | 1 Tell me please wherein may be trouble? Full log below ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/location ERROR: db_cassandra [dbcassa_table.c:592]: dbcassa_read_table_schemas(): Full dir name= /usr/local/etc/kamailio/cassandra/version INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.allow INFO: permissions [permissions.c:608]: mod_init(): default allow file (/usr/local/etc/kamailio//permissions.allow) not found => empty rule set INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /usr/local/etc/kamailio//permissions.deny INFO: permissions [permissions.c:617]: mod_init(): default deny file (/usr/local/etc/kamailio//permissions.deny) not found => empty rule set INFO: misc_radius [misc_radius.c:164]: mod_init(): initializing... ERROR: db_cassandra [dbcassa_base.cpp:449]: cassa_translate_query(): table version does not exist! ERROR: db_cassandra [dbcassa_base.cpp:962]: db_cassa_query(): Failed to query Cassandra cluster ERROR: <core> [db.c:401]: db_table_version(): error in db_query ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table location ERROR: usrloc [dlist.c:640]: register_udomain(): error during table version check. ERROR: registrar [reg_mod.c:554]: domain_fixup(): failed to register domain ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:982 ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:987 ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio//kamailio.cfg:992

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Kamailio + Asterisk with multidomain
by Tomas Zanet 07 Feb '17

07 Feb '17

Hello, thanks to this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I successfully installed Asterisk 11.6 and Kamailio 4.4.4 on the same machine. Everything works fine. Right now I would like to extend this scenario adding MULTIDOMAIN support, which is not enabled by default Do you see any problems / technical limitations to add multidomain support in this scenario? Before enhancing Kamailio cfg file and Asterisk configuration I would like to know: is this possible? As far as I know Kamailio works fine with multidomain support but I don’t know if Asterisk or both processes can support it. Thanks in advance Regards, T. Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 1

DMQ Cluster - hosts loose network connectivity
by José Seabra 07 Feb '17

07 Feb '17

Hello there, My DMQ cluster has 4 nodes and by some reason 2 of them lost the network connectivity for long time(~ 4 hours), after the network of these 2 nodes come back they didn't get connected on DMQ cluster automatically, i had to restart kamailio to get them again on DMQ list. My doubts here are: - Is this an expected behavior of DMQ module? - Is there any way of put them again on DMQ bus without need restart kamailio? Thank you Best Regards José Seabra

3 5

Kamailio parse failed message
by Hai Bui Duc Ha 07 Feb '17

07 Feb '17

Hi all, I have problem when make call with my Android mobile use PJSIP library. Scenario: my client -> Kamailio -> Freeswitch (media server) -> another client (soft phone on Windows) my client: + use Bluestack + Capture via Wireshark + use Wifi Issue: The call will be drop after ~ 30 second. I see the error on Kamailio: *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 13)* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<p:8 PCMA/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-16#015#012ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0#015#012Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias#015#012Max-Forwards: 70#015#012From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO#015#012To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj#015#012Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB#015#012CSeq: 29055 ACK#015#012Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>#015#012Content-Length: 0#015#012#015#012>* *Jan 5 16:08:59 ab-kz-02 kamailio[6343]: ERROR: <core> [receive.c:129]: receive_msg(): core parsing of SIP message failed (49.156.54.54:50785/2 <http://49.156.54.54:50785/2>)* Seem to the server error when parse (on INVITE SDP) *a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16* (on new ACK message) *ACK sip:buiduchahai@125.212.212.36:11000;transport=tcp SIP/2.0* *Via: SIP/2.0/TCP 10.0.2.15:57735;rport;branch=z9hG4bKPjlgc13AjrUrFJHq60vWhGqsUaGXi2F98Z;alias* *Max-Forwards: 70* *From: "Phap Huynh" <sip:huynhngocphap@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Ahuynhngocphap(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO* *To: <sip:buiduchahai@happy.anttel-pro.ab-kz-02.antbuddy.com <sip%3Abuiduchahai(a)happy.anttel-pro.ab-kz-02.antbuddy.com>>;tag=2SF4D790Zy6Kj* *Call-ID: ftMudIpIQeKWwP8kQDi2z1S0D1sV3KaB* *CSeq: 29055 ACK* *Route: <sip:125.212.212.40;transport=tcp;lr;ftag=zJBNvD67y3E.1I5Y5ZrRI4JmP5JKeNWO>* *Content-Length: 0* I think the SIP message is fragmented but when resume package is not correct. Do you have any advice ? Thank you for watching ! Regards, Hai Bui -- Hai Bui VoIP engineer, Cvoice team, HTK-HCM Office Mobile: +84-165-618-9876

2 11

FOSDEM 2017
by Daniel-Constantin Mierla 07 Feb '17

07 Feb '17

Hello, another edition of FOSDEM is approaching, about 2 weeks left: - https://fosdem.org/2017/ Couple of devs and many Kamailio friends will be around. Like past year, there is a Realtime Communications devroom, this edition on Sunday. Very interesting presentation for our ecosystem: Olle giving a presentation about IPv4/IPv6, Inaki introducing its SFU media server project, Jose talking about JsSIP, Daniel debating about fundraising FreeRTC, Lorenzo with Janus SIP-WebRTC gateway, the other Lorenzo with Homer Sipcature, Saul with Jitsi, Dan with cgrates, Matt with Asterisk, Giovanni with FreeSwitch... Schedule at: - https://fosdem.org/2017/schedule/track/real_time_communications/ On Sunday morning, part of Lua devroom, I will present about using Lua for building RTC services with Kamailio: - https://fosdem.org/2017/schedule/track/lua/ - https://fosdem.org/2017/schedule/event/luartcserviceskamailio/ During the past editions we organized a dinner on Saturday evening. Shall we attempt to do it again in advance this year? Or should we do it on spot based on the weather and mood at the end of Saturday? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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kamailio proxy behind firewall
by JBF 07 Feb '17

07 Feb '17

Hello, we have the following Configuration for our kamailio installation (we are using TLS and not udp) (1) F5 Firewall (configured as message fowarding), opening a TLS server on the outside (2) SIP proxy, with a TLS server accessed by the F5 . The SIP proxy doesnt see the F5 TLS server (3) SIP registrar REGISTER works find We have the following issue on INVITE: A sends an INVITE to B. The Registrar patches the R-URI with the content of location, which contains the publi ip of the Device (because the device used stun) we force the routing from registrar to proxy by using t_relay (SIP_PROXY_IP) /The proxy tries to route to this R-URI, which is not visible/ I am not sure how to fix that: Record Route is for a true sip proxy, but the Firewall does not have an server facing the SIP proxy: the sip proxy needs to find the proper client socket opened at register to route the INVITE We have arranged for the Firewall to add its own Via, but if i understand correctly, this is used for replies, and here we are dealing with a request forwarding, and t_relay uses the r-ruri to route requests. IT might be why REGISTER works correctly (ie the 200 OK is routed correctly from proxy to firewall) I could arrange for the location table to contain the private ip and port of the firewall connection (through the use of the received/rport info inserted in the Via by the proxy ) That would mean, however that the contact of the user will contain the private interface of the F5 which i found weird. How do you think i should proceed ? any advices are welcome Thank you -- View this message in context: http://sip-router.1086192.n5.nabble.com/kamailio-proxy-behind-firewall-tp15… Sent from the Users mailing list archive at Nabble.com.

2 3

RTPEngine disable/enable crash kamailio
by Sergey Basov 07 Feb '17

07 Feb '17

Hi All. I ave just try to temporary disable 1 of 2 RTPEngines. It was disabled successfully all media dialogs was finished. But when I have try to enable it I have kamailio crashed with error: CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 121 Does anyone have such issue? kamailio 4.4.5 on RHEL 7.2 x86_64 rtpengine 4.4.0 on RHEL 7.2 x86_64 # kamctl fifo nh_show_rtpp all which: no gdb in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin:/root/bin:/) udp:10.1.23.24:2223:: set=0 index:: 0 disabled:: 0 weight:: 1 recheck_ticks:: 0 udp:10.1.23.25:2223:: set=0 index:: 1 disabled:: 0 weight:: 1 recheck_ticks:: 0 after # kamctl fifo nh_enable_rtpp udp:10.1.23.25:2223 0 it was successfully disabled with media going on after # kamctl fifo nh_enable_rtpp udp:10.1.23.25:2223 1 I got kamailio crash with error CRITICAL: <core> [pass_fd.c:277]: receive_fd(): EOF on 121 sorry I does not have coredump, it is production system. my config regarding rtpengine: modparam("rtpengine", "rtpengine_allow_op", 1) modparam("rtpengine", "rtpengine_sock", "udp:10.1.23.24:2223 udp:10.1.23.25:2223") Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

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