sr-users November 2018

sr-users@lists.kamailio.org

68 participants
92 discussions

IMS: SMS over IP - how to get the originator's tel address?
by Ron McLeod 22 Nov '18

22 Nov '18

I am working on a VoLTE/IMS scenario to transfer text messages as SMS over IP (+g.3gpp.smsip feature) between UE's. This is a high-level view of the message flows between the endpoints (not showing CSCFs): UE1 IP-SM-GW |---- MESSAGE (SUBMIT) --------------->| |<------------------- 202 ACCEPTED ----| | | |<-------- MESSAGE (SUBMIT_REPORT) ----| |---- 200 OK ------------------------->| | UE2 | | | |<-------------- MESSAGE (DELIVER) ----| |---- 200 OK ------------------------->| The messages does get transferred from UE1 to UE2 successfully. Since this is GSM-style SMS, the OA (originating address) and DA (destination address) are specified using telephone numbers, not sip addresses. This is not a problem for the DA, since it is specified in the SUBMIT TPDU from the originating handset, but when the IP-SM-GW sends the DELIVER TPDU it needs to specify the OA so that the recipient knows who sent the SMS, and this was not provided in the MESSAGE with the SMS-SUBMIT. According to 3GPP TS 24.341 (http://www.qtc.jp/3GPP/Specs/24341-b20.pdf): "The IP-SM-GW will have to use an address of the SM-over-IP sender that the SC can process (i.e. an E.164 number). This address will come from a tel URI in a P-Asserted-Identity header (as defined in RFC 3325 [13]) placed in the SIP MESSAGE request by the P-CSCF or S-CSCF." Looking at the MESSAGE request sent from the S-CSCF to the IP-SM-GW, it looks like neither the P-CSCF nor the S-CSCF is providing the tel address in a P-Asserted-Identity header: MESSAGE tel:+3108012345680 SIP/2.0 Route: <sip:defaultapp@tas.core.ims1.test;lr>, <sip:iscmark@scscf.core.ims1.test:6060;lr;s=1;h=0;d=0;a=7369703a303031303330 31323334353637383940696d732e6d6e633030332e6d63633030312e336770706e6574776f72 6b2e6f7267> P-Served-User: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org>;sescase=orig;regstat e=reg f: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org>;tag=2703358633 t: <tel:+3108012345680> CSeq: 555874977 MESSAGE i: 2703358625_2324045032@2001:470:eb88:b021:27f:9507:589d:c201 Via: SIP/2.0/UDP [2001:470:EB88:150:0:0:0:22]:6060;branch=z9hG4bKb857.bd3345ba81ec10b1d11639d 9c7414606.0;i=a Via: SIP/2.0/TCP [2001:470:EB88:150:5054:FF:FE44:BFC9];branch=z9hG4bKb857.be53cc1c583b9c5f9dc ebf3306f2b7c1.0 v: SIP/2.0/UDP [2001:470:eb88:b021:27f:9507:589d:c201]:8906;rport=8013;branch=z9hG4bK355262 2488 Max-Forwards: 68 P-Access-Network-Info: 3GPP-E-UTRAN-FDD; utran-cell-id-3gpp=0010300030000001 c: application/vnd.3gpp.sms Allow: MESSAGE d: no-fork User-Agent: Xiaomi_Redmi 3S_Android6.0.1_9 l: 27 P-Charging-Vector: icid-value=49565300000000511100004B01000000; icid-generated-at=2001:470:EB88:150:5054:FF:FE44:BFC9 P-Asserted-Identity: <sip:001030123456789@ims.mnc003.mcc001.3gppnetwork.org> P-Visited-Network-ID: core.ims1.test Message Body GSM A-I/F RP - RP-DATA (MS to Network) Message Type RP-DATA (MS to Network) RP-Message Reference RP-Message Reference: 0x05 (5) RP-Originator Address RP-Destination Address - (3108012345680) RP-User Data Length: 14 TPDU (not displayed) GSM SMS TPDU (GSM 03.40) SMS-SUBMIT 0... .... = TP-RP: TP Reply Path parameter is not set in this SMS SUBMIT/DELIVER .0.. .... = TP-UDHI: The TP UD field contains only the short message ..1. .... = TP-SRR: A status report is requested ...0 0... = TP-VPF: TP-VP field not present (0) .... .0.. = TP-RD: Instruct SC to accept duplicates .... ..01 = TP-MTI: SMS-SUBMIT (1) TP-MR: 92 TP-Destination-Address - (8000) TP-PID: 0 TP-DCS: 0 TP-User-Data-Length: (5) depends on Data-Coding-Scheme TP-User-Data SMS text: HELLO Is this something that in configurable at the P-CSCF or the S-CSCF, or would it require now development? Thanks, Ron

2 3

issues with the debian repo for kamailio 5.2
by Aleksandar Sosic 22 Nov '18

22 Nov '18

Hi guys, I was trying to install the last 5.2 kamailio version editing the `/etc/apt/sources.list`: ``` deb http://deb.kamailio.org/kamailio52-nightly xenial main deb-src http://deb.kamailio.org/kamailio52-nightly xenial main ``` But as you can see there's only the 5.3 and 4.3 available: ``` root@e7845b3041eb:/# apt-cache madison kamailio kamailio | 5.3.0~dev0+0~20181114003826.5+xenial | http://deb.kamailio.org/kamailio52-nightly xenial/main amd64 Packages kamailio | 4.3.4-1.1ubuntu2 | http://archive.ubuntu.com/ubuntu xenial/universe amd64 Packages kamailio | 4.3.4-1.1ubuntu2 | http://archive.ubuntu.com/ubuntu xenial/universe Sources kamailio | 5.3.0~dev0+0~20181114003826.5+xenial | http://deb.kamailio.org/kamailio52-nightly xenial/main Sources ``` Any hints on how to solve this are very welcome. :) Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

5 6

t_set_retr and NOTIFY send after handle_publish
by Vasiliy Ganchev 21 Nov '18

21 Nov '18

Hi, list! I have a tricky problem, need advise from the community. The problem I need to solve: - have a SIP device, with a big number of BLFs configured, in some moment of time the device is slow to handle incoming SIP messages (a lot of NOTIFYs received), kamailio start retransmitting NOTIFYs, and the device get stuck. My idea is to change retransmits timeout for NOTIFYs that are sent out after handle_publish, so if get stuck, not to make "DoS" I have. What I tried: - t_set_retr(5000, 10000); before handle_publish => does not work - my tm is compiled with DTM_DIFF_RT_TIMEOUT, and t_set_retr works for e.g. INVITEs I have read the mailing list and also studied sources of presence and tm modules. I see that at the moment there is no way to solve what I want from config. My next steps are going to make some hacks in C code... at least to understand the code better and find some new insights Any other ideas? Comments/ideas are highly welcomed! cheers -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 3

Closing all ws connections
by Andrey Deykunov 21 Nov '18

21 Nov '18

Hi guys, I've implemented an rpc call that closes all WS connections: void ws_rpc_close_all(rpc_t *rpc, void *ctx) { ws_connection_t **list = NULL, **list_head = NULL; ws_connection_t *wsc = NULL; int ret; list_head = wsconn_get_list(); if(!list_head) return; list = list_head; wsc = *list_head; while(wsc) { LM_WARN("Closing connection\n"); ret = close_connection(&wsc, LOCAL_CLOSE, 1000, str_status_normal_closure); wsc = *(++list); } wsconn_put_list(list_head); } Please take a look. Can I call close_connection() and perform list iterations in this function not using WSCONN_LOCK? Thanks, Andrey

3 2

from Asterisk to Kamailio
by Valter Nogueira 21 Nov '18

21 Nov '18

Today, I use Asterisk as a SIP/RTP PROXY I proxy from customers Asterisks to a VOIP provider, in a multi-homed server. Now, I want to move to Kamailio without any rupture in customer's configuration. As anyone can imagine I am kind of lost. USER ACCOUNTS In Asterisk, I create a dynamic host account named ACCOUNT1 and I receive in *FROM HEADER sip:ACCOUNT1@customer_ip_address* In Kamailio, I have to define the account's domain like *kamctl add ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com> password. *Kamailio just accepts a REGISTER/INVITE from *ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com>* SIP/RTP PROXY In Asterisk, I just dialout to the VOIP PROVIDER like *dial (SIP/VOIP_ACCOUNT/${EXTENSION})* Asterisk does all the magic (it is a B2BUA). It bridges the new call and media to the original call. Moreover, user don't know anything about how call are completed, nor how credentials are setup and soon. In Kamailio, I guess that I have to use nat, tm and rtpproxy modules and maybe uac. I am not sure how to setup it. Can someone send me a clue? Thank you, Valter

4 7

Kamailio-5.1.6 Crash
by Hamid Hashmi 20 Nov '18

20 Nov '18

I have been using the following version of kamailio Version: kamailio 5.1.6 (x86_64/linux) Default config: /usr/local/etc/kamailio/kamailio.cfg Default paths to modules: /usr/local/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: unknown Compiled with: gcc 4.8.5 Compiled on: 11:48:33 Oct 11 2018 and I noticed today that the Kamailio service is crashing. [root@kamailiostandby /]# gdb core.17805 GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-100.el7_4.1 Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... [New LWP 17805] Core was generated by `/usr/local/sbin/kamailio -P /run/kamailio/kamailio.pid -f /usr/local/SIP'. Program terminated with signal 11, Segmentation fault. #0 0x00007f636559e54b in ?? () "/core.17805" is a core file. Please specify an executable to debug. (gdb) list No symbol table is loaded. Use the "file" command. (gdb) bt #0 0x00007f636559e54b in ?? () #1 0x00007f63a9571810 in ?? () #2 0x0000000000000000 in ?? () Regards Hamid R. Hashmi

2 3

registrar::save() does not save 'received' field of the 'location' table
by Andrey Deykunov 20 Nov '18

20 Nov '18

Hi, We deploy kamailio in the cluster containing two nodes (Active and Passive). When Active node becomes Passive, our websocket clients continue sending REGISTER to this node. Passive node routes these requests to the Active node through the UDP, that causes the following warning message: 2018-11-14T04:43:31.070005-08:00 qapbx-ca1-1 kamailio[25597]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:64.78.52.88:5060, to ws:10.9.3.116:20033) Every registered client should be stored in the 'location': $var(save_val) = save("location", 0, "sip:$au@$td"); if ($var(save_val) <= 0) { xlog("L_ERR", "[$fU,R,$ci]: Registration saving failed, contact $ct"); sl_reply_error(); } else if ($var(save_val) == 3) { xlog("L_INFO", "[$fU,R,$ci]: Unregistration accepted. Login: $fU; ua: $ua; s-ip: $si; s-port: $sp; 3d parm: sip:$au@$td"); } else { xlog("L_INFO", "[$fU,R,$ci]: Registration accepted. Login: $fU; ua: $ua; s-ip: $si; s-port: $sp; 3d parm: sip:$au@$td"); } Log output of the Avtive node: 2018-11-14T04:43:31.063376-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: REGISTER received with R-URI: sip:64.78.52.90, ct: <sip:vu2o757j@si7ofk7i8cs7.invalid;transport=ws>;+sip.ice; reg-id=1;+sip.instance="<urn:uuid:00ff4c8d-2a5e-4614-b80a-d7c14c0e41e3>";expires=600, cseq=1, s-ip: 64.78.52.88, s-port: 5060 2018-11-14T04:43:31.063947-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: Our node is active 2018-11-14T04:43:31.064323-08:00 qapbx-ca1-2 kamailio[24722]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: Authorization headers are empty in REGISTER command. Send 401 Unauthorized to 64.78.52.88 2018-11-14T04:43:31.285967-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: REGISTER received with R-URI: sip:64.78.52.90, ct: <sip:vu2o757j@si7ofk7i8cs7.invalid;transport=ws>;+sip.ice; reg-id=1;+sip.instance="<urn:uuid:00ff4c8d-2a5e-4614-b80a-d7c14c0e41e3>";expires=600, cseq=2, s-ip: 64.78.52.88, s-port: 5060 2018-11-14T04:43:31.286564-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: Our node is active 2018-11-14T04:43:31.286943-08:00 qapbx-ca1-2 kamailio[24716]: INFO: <script>: [54004221,R,ibca3ujmr0q81nn0l5puen]: Registration accepted. Login: 54004221; ua: JsSIP 3.2.11; s-ip: 64.78.52.88; s-port: 5060; 3d parm: sip:54004221@qapbx-ca1.qaserverdata.net but in this scenario the save() method stores null value in the 'received' row. How can I fix it? Thank you, Andrey

2 1

RTPengine: shouldn't SDP <sess-version> be incremented when SDP is updated in-dialog?
by George Diamantopoulos 20 Nov '18

20 Nov '18

Hello all, I have the following issue (I think) with RTP engine. An INVITE comes in, and rtpengine will rewrite the SDP accordingly, as configured in kamailio.cfg. After some time a reINVITE is sent out in the opposite direction, for session refresh purposes. As I use rtcp-mux-offer in kamailio.cfg for this direction, RTPengine will inject the rtcp-mux parameter, and this reINVITE is forwarded to the UAC that sent the original INVITE. However, the SDP in the reINVITE is exactly the same as the SDP in the 200 OK to the original INVITE, with the exception of the rtcp-mux parameter. Since the SDP offered from the same end has changed, shouldn't session version be incremented as well? Does this sound like something that should be reported as a bug to rtpengine? Or am I missing something here? Thanks! Best regards, George Diamantopoulos

3 4

Number of sql connections
by Duarte Rocha 19 Nov '18

19 Nov '18

Greetings, I have my core parameter listen like this : listen=udp:eth0 children=12 Like this and the other settings, my kamailio has 21 threads and 20 Sql connections. If I add more listens, listen=udp:eth0 listen=udp:eth0:5072 listen=udp:eth0:5074 Now I have 45 threads and 45 Sql connections. Why does this happen and can i set a a number of threads and/or sql connections that doesn't change with the number of "listen" parameters that i have ? Best Regards, Duarte Rocha

3 2

recording audio stream
by bill＠telopar.net 18 Nov '18

18 Nov '18

Hi All: I have a project that requires that I record the audio streams from an ongoing phone call. It seems that rtp_proxy module should do that, but I can't seem to get it to work. My config file: *if(nat_uac_test("8")) { //rtpproxy_manage("co"); xlog("L_INFO", "rtpproxy Call is going to private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("crwie","10.1.1.2"); } else { //rtpproxy_manage("cor"); xlog("L_INFO", "rtpproxy Call is coming from a private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("crwei","10.1.1.2"); } *There appears to be conversations going on with rtpproxy on both legs of the call. But I seem to be missing the step that records the audio*. *Anybody have an example of how to do this. TIA, Bill **

4 3

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sr-users November 2018