sr-users February 2014

sr-users@lists.kamailio.org

114 participants
158 discussions

Increasing MAX_RECV_BUFFER_SIZE
by Wingsravi R 09 Feb '14

09 Feb '14

Dear Kamailio'ns I am working on Kamailio (V 4.0)+ Mediaproxy server (2.5.2) running on Ubuntu 12.04. I am experiencing some Latency, jitter, pixelled, choppy video errors in my set-up. So i wanted to increase buffer sizes of my kamailio server , Mediaproxy server and Ubuntu also. In that regards i have increased Shared memory value to 512 MB and Package memory value 16 MB. BUFFER SIZE value to 8 MB in *config.h* file, But where can i change the value for 'MAX_RECV_BUFFER_SIZE' of kamailio? And are there any settings to increase Buffer Size of Mediaproxy server ? Is this changes will affect on my issues ? PS: I even changed ubuntu settings also : net.core.rmem_max= 8388608 net.core.wmem_max= 8388608 (but these values reset to default values again when i restarted system.) Please help me in resolving this issues. Any help will appreciate. Regards, Ravi

4 4

kamailo uses wrong ip address
by jaflong jaflong 07 Feb '14

07 Feb '14

I have this in my config $ru = "sip:"+$rU+"10.31.2.101:5090;transport=tls"; if (!t_relay_to_tls()) { sl_reply_error(); return; } My ip address dest is 10.31.2.101. Why has 130.31.2.101 been used? 13(10882) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 130.31.2.101 4(10883) ERROR: <core> [tcp_main.c:4422]: tcpconn_main_timeout(): connect 130.31.2.101:5090 failed (timeout)

2 1

My Kamailio + Asterisk != NOT realtime integration: TRANSFERERNAME and BLINDTRANSFER
by Alexandr Usov 07 Feb '14

07 Feb '14

Integration - works. Problem - dialing peer to peer via Kamailio OK but with missing VARs and extension number, on dialing/transferring. Maybe you know other way to configure Asterisk dialplan for users, registered on kamailio and alowing dial as SIP/UserNumber insted SIP/ UserNumber(a)kamailio.host.name What we have. My Kamailio config: *http://pastebin.com/p7YxsFaw <http://pastebin.com/p7YxsFaw>* Asterisk's user (peer) - registered on kamailio: [3] host=192.168.144.212 qualify=yes dtmfmode=rfc2833 canreinvite=no context=local-routing host=dynamic type=friend directmedia=no nat=no qualify=yes disallow=all allow=ulaw allow=alaw allow=g729 call-limit=2 limitonpeers=yes callcounter=yes callerid=Usov Mob <3> ### Asterisk queue members: ;syntax: member => interface,[,penalty][,membername][,state_interface][,ringinuse] member=SIP/1(a)sip.cloudpbx.com.ua,1,1001,SIP/1,no member=SIP/3(a)sip.cloudpbx.com.ua,1,1003,SIP/1,no ### Asterisk dialing local peer: ... exten => _X,n,Dial(SIP/${EXTEN}(a)sip.cloudpbx.com.ua,12,tT) ... ### Asterisk attended transfer from 1 to 9 exten: *http://pastebin.com/k9H4vMgx <http://pastebin.com/k9H4vMgx>* Problem #1: Peer 9 receive clid as asterisk(a)sip.cloudpbx.com.ua Need #1: 1(a)sip.cloudpbx.com.ua. Problem #2: TRANSFERERNAME=SIP/sip.cloudpbx.com.ua-000000b5 Need #2: TRANSFERERNAME=SIP/1(a)sip.cloudpbx.com.ua-000000b5 ### Asterisk blind transfer dump log: *http://pastebin.com/NXqXingR <http://pastebin.com/NXqXingR>* Problem #3: BLINDTRANSFER=SIP/sip.cloudpbx.com.ua-000000ba Need #3: BLINDTRANSFER=SIP/9(a)sip.cloudpbx.com.ua-000000ba Main problem it is missed peer number - SIP/*peer_number*@ sip.cloudpbx.com.ua. And asterisk(a)sip.cloudpbx.com.ua instead 1(a)sip.cloudpbx.com.ua on attended transfer. Asterisk sip.conf domain settings: realm=sip.cloudpbx.com.ua fromdomain=sip.cloudpbx.com.ua ;domain=sip.cloudpbx.com.ua ;; temporary not uses becaouse not accepting GOIP sim ports registration on asterisk with IP-address of sip proxy instead domain sip.cloudpbx.com.ua; and it's not helps to change asterisk(a)sip.cloudpbx.com.ua clid Any help from all - wellcome! P.S. domain sip.cloudpbx.com.ua not exist

1 1

CONFIRMED_NA dialog state timeout
by Alex Balashov 07 Feb '14

07 Feb '14

Hi, We frequently run into the situation where a call is simultaneously CANCEL'd by the caller and answered (2xx) by the callee. This results in the caller not sending an e2e ACK, since it CANCEL'd the branch. It results in retransmission of the 200 OK, since the proxy can't CANCEL the branch anymore. It leaves the dialog in state 3 #define DLG_STATE_CONFIRMED_NA 3 /*!< confirmed dialog without a ACK yet */ (dlg_hash.h) for up to 'dialog_timeout' time. Is there a straightforward hack for this situation? Does changing the dialog timeout AVP have any effect after the relay of the initial INVITE (and thus invocation of the registered dialog init TM callback)? More generally, I was wondering if there has been any thought given to the idea of separate timeout modparams for confirmed-nonACKed state, as well as early state perhaps. Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

ssl error
by jaflong jaflong 06 Feb '14

06 Feb '14

This is my tls.cfg for server [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/asterisk/certs/proxy.key certificate = /etc/asterisk/certs/proxy.crt As far as I understand (verify_certificate = no), and (require_certificate = no) should allow a client connecting without certicates. Can anyone understand what this debug indicates What is causes this error tls_read_f(): TLS read:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca 7(14841) DEBUG: tls [tls_server.c:178]: tls_complete_init(): Using TLS domain TLSs<default> 7(14841) DEBUG: tls [tls_domain.c:609]: sr_ssl_ctx_info_callback(): SSL handshake started 7(14841) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 7(14841) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7eff24e41260 n=940 fd=5 7(14841) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= 7(14841) DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x8df1e0, 5, 2, 0x7eff24e41260), fd_no=1 7(14841) DEBUG: tls [tls_domain.c:621]: sr_ssl_ctx_info_callback(): SSL handshake done 7(14841) DEBUG: tls [tls_domain.c:624]: sr_ssl_ctx_info_callback(): SSL disable renegotiation 7(14841) DEBUG: tls [tls_server.c:348]: tls_accept(): TLS accept successful 7(14841) DEBUG: tls [tls_server.c:355]: tls_accept(): tls_accept: new connection from 10.1.2.3:35573 using TLSv1/SSLv3 AES256-SHA 256 7(14841) DEBUG: tls [tls_server.c:358]: tls_accept(): tls_accept: local socket: 10.1.2.4:6443 7(14841) DEBUG: tls [tls_server.c:369]: tls_accept(): tls_accept: client did not present a certificate 7(14841) DEBUG: <core> [tcp_main.c:2556]: tcpconn_do_send(): tcp_send: sending... 7(14841) DEBUG: <core> [tcp_main.c:2590]: tcpconn_do_send(): tcp_send: after real write: c= 0x7eff24e41260 n=250 fd=5 7(14841) DEBUG: <core> [tcp_main.c:2591]: tcpconn_do_send(): tcp_send: buf= 7(14841) DEBUG: <core> [tcp_read.c:296]: tcp_read_data(): EOF on 0x7eff24e41260, FD 5 7(14841) ERROR: tls [tls_server.c:1186]: tls_read_f(): TLS read:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca 7(14841) ERROR: <core> [tcp_read.c:1281]: tcp_read_req(): ERROR: tcp_read_req: error reading 7(14841) DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8df1e0, 5, -1, 0x10) fd_no=2 called 7(14841) DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7eff24e41260, state -2, fd=5, id=2 7(14841) DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7eff24e3c200 10(14844) DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7eff24e41260, -2 from 1 10(14844) DEBUG: tls [tls_server.c:597]: tls_h_close(): Closing SSL connection 0x7eff24e3c200

2 1

Dialog
by John Murray 06 Feb '14

06 Feb '14

Hi, I need to get call-id and from-tag from a call in ringing state (2). If I use dlg_manage() and put the call into a profile using set_dlg_profile("usr","1"). Then do a avp_db_query() I don't see the call until it is connected. I am using db_mode 1. Yet if I do a sercmd dlg.dlg_list I see the call in ringing state and when it is connected. How do I get the call-id when it is in ringing state? Regards John

1 0

Change of source ip address
by Diego Alejandro Ozuna Escalada 06 Feb '14

06 Feb '14

Dear list, I'm having a problem. I have an SBC server with two network interfaces, one external IP (X.X.248.194) and other internal IP (X.X.248.21). Messages are sent to the proxy sip server through internal ip with this configuration: route [sip-serverr] { rewritehostport ("ip_sip_proxy: 5060"); force_send_socket (X.X.248.21: 5060); t_relay (); exit; } When a UA sends an INVITE to the SBC, the responses (back) to UA are being sent with the correct external source IP (X.X.248.194) but if there are retransmissions of these responses, after 6 seconds, the source IP changes to the internal ip (X.X.248.21) Does anyone from the list know how I can solve this? Thanks, Regards, -- Diego

2 2

FS#380 - fixes for memory leak in TLS module
by Ding Ma 06 Feb '14

06 Feb '14

Daniel, Klaus, Sorry to reply late to your comments, was busy doing some other things until now. Just provided some comments in https://sip-router.org/tracker/index.php?do=details&task_id=380 Please take a look and let me know if you agree with my assessment. Thanks, Ding

1 0

bug in authdb_mod ??
by jay binks 06 Feb '14

06 Feb '14

So when Im trying to use is_subscriber & load_credentials together, It seems I have either hit a bug or just a bad config ( but I dont think its the latter ) My symptom is that only the field on the far right of "load_credentials" ever returns the correct value, the other fields are always null. My config has this setting : modparam("auth_db", "load_credentials", "field1;field2") and I can see the following debug when running kamailio -ddddd 0(14174) DEBUG: auth_db [authdb_mod.c:291]: w_is_subscriber(): uri [sip: 1112(a)192.168.173.5] table [subscriber] flags [1] 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:807]: db_cassa_query(): query table=subscriber 0(14174) DEBUG: db_cassandra [dbcassa_table.c:461]: dbcassa_db_get_table(): found dbname=kamailio, table=subscriber 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:351]: cassa_constr_key(): query col = username 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:351]: cassa_constr_key(): query col = domain 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:376]: cassa_constr_key(): key = 1112 192.168.173.5 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:499]: cassa_translate_query(): Query col: field2 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:499]: cassa_translate_query(): Query col: field1; field2 0(14174) DEBUG: db_cassandra [dbcassa_base.cpp:501]: cassa_translate_query(): get 2 columns 0(14174) DEBUG: <core> [db_res.c:118]: db_new_result(): allocate 56 bytes for result set at 0x7fc6ce770e78 if I had to put money on where the bug is, I would look in parse_aaa_pvs in authdb_mod.c however, an initial glance revealed nothing glaring, also this file was last changed in *2013-01-20 ( and that was only a move from modules_k )* SURELY I cant be the only person wanting to use this for more than one field ? This code looks like its almost the same as when first committed in August 2011 .... Id love someone to see if they can confirm this and point me in the right direction ( if this is my fault ) or lets get this fixed, if its a bug :) -- Sincerely Jay

2 5

Handling RFC3578 (Overlap Dialing) in Kamailio/ Asynchronious transaction handling
by Moritz Graf 06 Feb '14

06 Feb '14

Hoho, i want to implement RFC3578 (transaction oriented) in Kamailio. Therefore I have to correlate two different Invites with same Call-ID. I see various ways of accomplishing this, but I am stuck at all of my solutions. Find the algorithm I want to implement attached. Shortly explained what RFC3578 is: In a open numbering plan you never know if the INVITE you received is already complete, or if there are more numbers coming in. One way of accomplishing this is to set up a timer. If the timer elapses you assume the number is complete. If not, you are receiving a new INVITE with one digit more. Now you have to close the old transaction with a "484 - Address Incomplete"-response and start the timer again. (Find the algorithm I want to implement attached) The following lines, I'm going to explain my thoughts. For reference I'm calling "Thread A" the first INVITE coming in and "Thread B" the second. ####################### ### using Module ASYNC * Thread A -> async_route("handle_overlap","<timer t99 value>"); * Thread B -> if new incoming transaction, ** set avp("<call-id>")= <digitlength of called number> and ** call async_route("handle_overlap","<timer t99 value>"); * in route[handle_overlap] ** if avp("call-id") greater than "<digitlength of called number>" *** cancel transaction with "484 -Address Incomplete" ** if avp("call-id") equals "<digitlength of called number>" *** handle call => the number is complete PROBLEM: The old transaction is not canceled immediately, but after t99 elapsed... ####################### ### using Module TMX using t_suspend() and t_continue(), but here i see the following problems: * Thread A calls "t_suspend()" ** How is it possible to cancel "Thread A"'s transaction from Thread B? ** If there is no Thread B, so the numbers were complete, how can I get Thread A to continue processing?? * General TMX-Question: Are there any examples available? ####################### ### using simple sleep() and shv()/avp() Sounds like dirty hacking... I'd appriciate a solution with one of aboves modules. But if there is a solution available, I'd try this. Thx in advise Moritz -- Moritz Graf, B.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg Telefon +49 (9 41) 69 85 - 1 86 Telefax +49 (9 41) 69 85 - 2 86 mailto:moritz.graf@g-fit.de http://www.g-fit.de G-FIT Gesellschaft für innovative Telekommunikationsdienste mbH & Co. KG, Kommanditgesellschaft, Sitz Regensburg, Registergericht Regensburg, HRA 7626; Geschäftsführer: Dipl.Inf. (FH) Alfred Rauscher

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sr-users February 2014