sr-users January 2020

sr-users@lists.kamailio.org

80 participants
95 discussions

WebRTC ACK Protocol
by Ilie Soltanici 08 Jan '20

08 Jan '20

Hello, I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine. So far, everything is working fine, I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. This is how invite is looking: INVITE sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss SIP/2.0 Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060> Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE User-Agent: Proxy Date: Wed, 03 Apr 2019 17:11:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Alert-Info: Content-Type: application/sdp Content-Length: 596 Server: SIP Proxy and then WebRTC app is replying with 200 OK: SIP/2.0 200 OK Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE Contact: <sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Proxy-WEBRTC Content-Type: application/sdp Content-Length: 901 and finally, Kamailio is trying to send this ack through UDP protocol: ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0 Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport Route: <sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes> Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 ACK User-Agent: Proxy Content-Length: 0 If i'm trying to force it through TLS, i'm receiving error: get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443, to udp:22.22.22.22:23317) Can someone point me in the right direction, please? Thank you.

3 4

Kamailio when acting as client doesnt send SNI in client hello handshake message
by mahesh b 07 Jan '20

07 Jan '20

Hi, Am using Kamailio 5.1.9 version My Setup : client1 -> kamailio server 1 ( IP : 10.211.160.172) ----> kamailio server 2( IP : 10.211.160.176) -> client2 I have a scenario where kamailio server 1 has to initiate an outgoing tls connection to kamailio server 2, i have set the server_name and server_id in the client profile in tls.cfg like below on kamailio server 1 [client:default] verify_certificate = no require_certificate = no server_name = mahesh.client.com [client:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile2/btip_172_server_private.key certificate = /root/mahesh_openssl/profile2/btip_172_server_public.crt ca_list = /root/mahesh_openssl/profile2/btip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = btip.176.com server_id = btip.176.com And in sar.cfg $xavp(tls=>server_name)="btip.176.com"; $xavp(tls=>server_id)="btip.176.com"; $du = "sip:10.211.160.176:5061;transport=tls"; .... t_relay(); What i observe is that , when client hello is sent by 10.211.160.172 to 10.211.160.176, i dont see Extension server_name being sent. Am i missing anything. Please help !

4 7

app_lua with kamailio 5.3.X
by Aymeric Moizard 07 Jan '20

07 Jan '20

Hi All! I have moved from kamailio 5.2.X to 5.3.X and I have now an error with calling "sr.err" from my lua script: sr.err("ERROR: param $tU [" .. tu .. "]\n") app_lua [app_lua_api.c:626]: app_lua_runstring(): error from Lua: /etc/kamailio/lua/my-script.lua:69: attempt to index global 'sr' (a nil value) I do understand that I should add this to continue with old way: loadmodule "app_lua_sr.so" modparam("app_lua_sr", "register", "sl") However, I would prefer to fix my script to use the NEW way but can't find it. Thanks a lot. Aymeric -- Antisip - http://www.antisip.com

2 1

RTPEngine performance - How far are you pushing your systems
by davy van de moere 07 Jan '20

07 Jan '20

Dear all, Not a direct question on kamailio, but on rtpengine, so feel free to ignore :) I have a farm of rtpengine servers , supporting our set of kamailio's. On recent decent hardware (hp dl360 gen9/10 servers), I typically push my systems upto 2500 channels per machine. Then I tend to see the load of the machine go up in a logarithmic manner. E.g. a load of 3 supports 2500 channels, but when its at 3000 channels, load can be at 8. etc. I have the kernel module running, and I have a good impression it's being used too. No transcoding is going on, no recording, ... I've tried on different OS'es, different network cards, and I always end up at that 2500 channels +-. Is that what I can expect from rtpengine? Or am I really overlooking something? What numbers do you push out of your systems? kind regards and happy 2020!

5 10

Siremis & freeswitch
by John Tuxies 07 Jan '20

07 Jan '20

I have a Debian server running Kamailio and i have all my users regiatering and talking to each other. I do create/remove users from Siremis. I have followed the guide from here: https://www.powerpbx.org/content/kamailio-v5-siremis-v5-debian-v9-mariadb-a… I would like to add freeswitch to the same server and add conference, voicemail, supplementary services for my users. I have given it a try to install freeswitch and tried to configure it accordingly to https://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc But.now the Siremis gui shows a lot of errors and i cannot login any more. Since it is a Virtual machine i reverted it back and it works without freeswitch. How could i integrate freeswitch to my system please?

4 11

opensource VoLTE/IMS Clients like IMSDroid
by Hamid Hashmi 06 Jan '20

06 Jan '20

Dear Users Can you please name some of VoLTE/IMS Clients (Like IMSDroid for Android) for Windows/Linux/MAC to test IMS Network. IMSdroid available clients on the internet are not working with android v8,9 and 10. or if someone can guide me to create a working APK from IMSDroid available code on Github<https://github.com/DoubangoTelecom/imsdroid>. Regards Hamid

1 0

SNI Query
by mahesh b 06 Jan '20

06 Jan '20

Hi All, Am using Kamailio 5.1.9 version. *Below is my tls.cfg* [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = server.key certificate = server.crt ca_list = bundle.crt cipher_list = RSA verify_depth = 9 [client:default] verify_certificate = no require_certificate = no [server:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile2/btip_172_server_private.key certificate = /root/mahesh_openssl/profile2/btip_172_server_public.crt ca_list = /root/mahesh_openssl/profile2/btip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = btip.172.com [server:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile1/ctip_172_server_private.key certificate = /root/mahesh_openssl/profile1/ctip_172_server_public.crt ca_list = /root/mahesh_openssl/profile1/ctip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = ctip.172.com My Kamailio server ip is 10.211.160.172 i)When i initiate a tls connection from remote server(which is also a kamailio server) say 10.211.160.176 to 10.211.160.172 In the client hello am setting sni name as btip.172.com => so on 10.211.160.172 side it is picking up the server profile with serve_name btip.172.com for the tls handshake.*// Working as expected* ii)When i initiate a tls connection from another remote server(Which is also a kamailio server) say 10.211.160.163 to 10.211.160.172 In the client hello am setting sni name as ctip.172.com => so on 10.211.160.172 side it is picking up the server profile with serve_name ctip.172.com for the tls handshake.*// Working as expected* iii)When i initiate a tls connection from another remote server(Which is also a kamailio server) say 10.211.160.175 to 10.211.160.172 In the client hello am NOT setting sni name => so on 10.211.160.172 side should it pick up the server default profile or the first profile to which IP and port matches ? what i observe from logs is that it is picking up the server profile with server_name ctip.172.com for the tls handshake. I had a look at the code in function tls_lookup_cfg, from the debug prints i understand it is trying to match profile for IP and port if ((p->port==0 || p->port == port) && ip_addr_cmp(&p->ip, ip))* // IP and port matched* { if(sname && sname->len>0) *//Incoming Client hello dint have sname, so it will hit the else part* { if(p->server_name.s && p->server_name.len==sname->len && strncasecmp(p->server_name.s, sname->s, sname->len)==0) { LM_DBG("socket+server_name based TLS server domain found\n"); return p; } } else { return p; *// so it is returning the first profile to which IP and port matched.* } } Am i missing anything or is this a bug ? if in the clienthello there is no sni , what needs to be done to make use of the default profile for the tls handshake ? Or is this something fixed in latest. I just Tried and Modified the code as below, after which it is giving the server default profile when no sni in Incoming Client Hello. if ((p->port==0 || p->port == port) && ip_addr_cmp(&p->ip, ip)) { if(sname && sname->len>0) { if(p->server_name.s && p->server_name.len==sname->len && strncasecmp(p->server_name.s, sname->s, sname->len)==0) { LM_DBG("socket+server_name based TLS server domain found\n"); return p; } } else { if( (type & TLS_DOMAIN_SRV) && (p->server_name.s) ) { LM_DBG("Inside %s at %d\n",__FUNCTION__,__LINE__); return cfg->srv_default; } else { LM_DBG("Inside %s at %d\n",__FUNCTION__,__LINE__); return p; } } } Regards, Mahesh.B

2 4

Installation issue
by Hassen Bencheikh 06 Jan '20

06 Jan '20

Good morning, I am a telecommunication engineer from the National Establishment of Aerial Navigation (Algeria). First of all i would like to thank you for making the SIP sever Kamailio available. But actually i have got issues during the installation process. I was using the steps described on the link below : https://www.kamailio.org/wiki/install/devel/git The issue was encountered at the step #make all, an error was found, and here is what was displayed on the terminal: *linux-vd3w:/usr/local/src/kamailio-devel/kamailio # make allmake -C src/ all make[1]: Entering directory '/usr/local/src/kamailio-devel/kamailio/src'generating core/autover.h ...CC (gcc) [kamailio] core/ver.oLD (gcc) [kamailio] kamailiowhich: no mysql_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mariadb_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mariadb_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)CC (gcc) [M db_mysql.so] my_res.oIn file included from my_res.c:24:0:my_cmd.h:27:10: fatal error: mysql.h: No such file or directory #include <mysql.h> ^~~~~~~~~compilation terminated.make[2]: *** [../../Makefile.rules:100: my_res.o] Error 1make[1]: *** [Makefile:511: modules] Error 1make[1]: Leaving directory '/usr/local/src/kamailio-devel/kamailio/src'make: *** [Makefile:30: all] Error 2linux-vd3w:/usr/local/src/kamailio-devel/kamailio #* Actually my knowledge on linux (opensuse leap) that i am using is very limited. Hopefully i will be able to use the server once installed and it will be used for ip recording tests purpose. Thank you for reading this email. Good bye.

5 4

MSRP proxy
by Sergey Safarov 06 Jan '20

06 Jan '20

I want proxy MSRP data on kamailio for SIP call (not websocket). To do this need update connection information for "message" media in SDP. How i can properly do it only for one media (other medias will be proxed using rtpproxy or rtpengine ). Looks as fix_nated_sdp fix for all medias in SDP. Sergey

2 1

pv_parse_spec2(): wrong char [)/41] in [$T(reply_reason)] at [15 (5)]
by Denys Pozniak 06 Jan '20

06 Jan '20

Hello! Looks like $T(reply_reason) does not work well (although $T_reply_reason is ok). Kamailio script: failure_route[TRUNK_FAILURE] { ... xlog("L_WARN","rm=$rm / $T(reply_reason) \n"); ... } 0(8894) ERROR: tmx [t_var.c:696]: pv_parse_t_name(): unknown PV name reply_reason 0(8894) ERROR: <core> [core/pvapi.c:952]: pv_parse_spec2(): pvar "T" has an invalid name param [reply_reason] 0(8894) ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in [$T(reply_reason)] at [15 (5)] 0(8894) ERROR: xlog [xlog.c:511]: xdbg_fixup_helper(): wrong format[rm=$rm / $T(reply_reason) https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables $T(name) - pseudo-variable class to access TM attributes The *name* can be: - reply_reason - reply reason [kamailio]$ kamailio -v version: kamailio 5.2.2 (x86_64/linux) 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 18:04:26 Apr 3 2019 with gcc 4.8.5 -- BR, Denys Pozniak

2 1

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sr-users January 2020