sr-users December 2016

sr-users@lists.kamailio.org

71 participants
104 discussions

kamailio dispatcher call flow example
by huan nguyen duy 23 Dec '16

23 Dec '16

Hi, I want to load balance multiple kamailio server using dispatcher module but not success at configuration. I’m not clearly how it work. Someone can show me an example call flow like below i’m very appreciated. Alice Proxy 1 Proxy 2 Bob | | | | | INVITE F1 | | | |--------------->| | | | 407 F2 | | | |<---------------| | | | ACK F3 | | | |--------------->| | | | INVITE F4 | | | |--------------->| INVITE F5 | | | 100 F6 |--------------->| INVITE F7 | |<---------------| 100 F8 |--------------->| | |<---------------| | | | | 180 F9 | | | 180 F10 |<---------------| | 180 F11 |<---------------| | |<---------------| | 200 F12 | | | 200 F13 |<---------------| | 200 F14 |<---------------| | |<---------------| | | | ACK F15 | | | |--------------->| ACK F16 | | | |--------------->| ACK F17 | | | |--------------->| | Both Way RTP Media | |<================================================>| | | | BYE F18 | | | BYE F19 |<---------------| | BYE F20 |<---------------| | |<---------------| | | | 200 F21 | | | |--------------->| 200 F22 | | | |--------------->| 200 F23 | | | |--------------->| | | | | Huan Nguyen, ——————————————————— Mobile app developer Phan Thi Company Sai Gon, VietNam

2 1

rtpproxy/rtpengine. Help to understand.
by Igor Olhovskiy 22 Dec '16

22 Dec '16

Hi! Issue I can’t figure out. Or all working ok and that’s just me who not understands. I have situation softPhone (111.222.3.2) -> Kamailio w. rtpproxy (111.222.3.3) -> PBX (111.222.3.4) All addresses are on same public network. rtpproxy is running with -l 111.222.3.3 -A 111.222.3.3 I want all media goes through rtpproxy. So, in INVITE and ACK I call rtpproxy_manage(). But according to sip packets on PBX (10.0.0.4) I got media address 10.0.0.2 in SDP, but not 10.0.0.3 If I use scheme, where client in some other NATted network (like 192.168.1.100), all is replaced correctly (means rtpproxy_manage() is working) So, I can’t understand, rtpproxy_manage() changes only rfc1918 addresses or I’m missing something? If any - can provide pcap’s with both cases. Regards, Igor

3 5

rtjson + mhomed and record route
by Diego Nadares 22 Dec '16

22 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } Thanks in advance. Diego.

1 0

Socket in rtjson module
by Diego Nadares 22 Dec '16

22 Dec '16

Hi Guys, Anybody knows if each 'socket' specified in routes struct should be use to send the messages? The socket is configured in kamailio. In kamailio.cfg listen=udp:172.16.213.38:5060 json struct "routes": [ { "uri": "sip:127.0.0.1:5080", "dst_uri": "sip:127.0.0.1:5082", "path": "<sip:127.0.0.1:5084>, <sip:127.0.0.1:5086>", ----> "socket": "udp:172.16.213.38:5060",<---- "headers": { "from": { "display": "Alice", "uri": "sip:alice@127.. In our kamailio has two eth and I want to select which one to use but I can't make it work through rtjson. In the code of the module I can't find where is being assigned the socket. Thanks in advance! Diego

2 3

kamailio dispatcher can not forward call in Amazon EC2
by huan nguyen duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information. Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

1 0

kamailio dispatcher can not forward call in Amazon EC2
by Huân Nguyễn Duy 22 Dec '16

22 Dec '16

1 0

Memory leak in Kamailio 4.4.4 TLS Module
by raja raja 22 Dec '16

22 Dec '16

Hi, shared memory not freed once TLS connection close means shm available memory decreasing for new connection that leads to the following error *tls: ssl bug #1491 workaround: not enough memory for safe operation* *TestCase:* 1. start kamailio 2. use sample program to establish tls connection and close imediately. 3. Kamailio Log is For first TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63604096 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63545192 For second TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63545192 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63489352 *Memory Leak:* for tls connection connection 63604096 - 63545192 = 58904 bytes not freed. for tls connection connection 63545192 - 63489352 = 55840 bytes not freed. I tested same scenario on kamailio - 4.2.4, 4.3.4 versions result is same. Could you please provide the fix? Thanks & Regards, Raja Chegondi.

2 1

register_timer cb function does not have access to shared memoery var
by Andy 22 Dec '16

22 Dec '16

in mod_init I have something like list = (struct list_t*) shm_malloc ..... this list is available from any functions in the module, now there is a timer registered, in the same module register_timer(callback_f, 0 , 5) the callback_f function when tries to use list, always sees as empty -- View this message in context: http://sip-router.1086192.n5.nabble.com/register-timer-cb-function-does-not… Sent from the Users mailing list archive at Nabble.com.

2 2

new URI [] shorter than old URI - error
by Nelson Migliaro 21 Dec '16

21 Dec '16

Hello, My SIP vendor request me to replace the from in order to authenticate calls. I use: uac_replace_from("111111111", "sip:111111111@vendor-domain.com"); And when finishing calls I have this error: ERROR: uac [replace.c:607]: restore_uri(): new URI [] shorter than old URI [sip:34912345678@vendor-public-ip] Information ------------------------------------------------------------------ 34912345678 -> original CLI 111111111 -> my vendor contract vendor-domain.com -> domain of the vendor ------------------------------------------------------------------ I use the function uac_replace_from in the request_route My Kamailio version is: 4.2.8-1.1 Thank you Nelson.-

1 0

kamailio 4.4 segmentation fault when dp_translate match dialplan with ftmatch operator and empty match expression
by Julia Boudniatsky 21 Dec '16

21 Dec '16

version: kamailio 4.4.4 id:852c47 ENTRY: { PRIO: 16765956 MATCHOP: 2 *MATCHEXP: * MATCHLEN: 0 SUBSTEXP: REPLEXP: ATTRS: 1 } Please see generated core file in extension. Thank you, Julia.

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sr-users December 2016