sr-users January 2020

sr-users@lists.kamailio.org

80 participants
98 discussions

Parsing exception when "user=phone" is present
by Duarte Rocha 08 Jan '20

08 Jan '20

Greetings, I'm testing some portability scenarios with carriers and i've ran into some issues. Take this example URI from ETSI TR 184 003 ( https://www.etsi.org/deliver/etsi_tr/184000_184099/184003/03.01.01_60/tr_18… ) - Sip:+49234598765;npdi;rn=+49-D123@provider5.net;user=phone In this case i need to parse user, user's parameters, domain and uri parameters. In order to that i'm using uri's transformations like {uri.user}, {uri.domain}, etc. However, when "user=phone" is present, Kamailio's parser works differently. Without "user=phone" -> {uri.user} is equal to "+49234598765;npdi;rn=+49-D123" With "user=phone" -> {uri.user} is equal to "+49234598765" . I really need to access "rn" and "npdi" parameters in those cases. How can i do it when "user=phone" is present? Is there a way to disable the "user=phone" exception for parsing? Best Regards, Duarte Rocha

2 2

possible TCP deadlock (tls again?) // pike module not releasing IPs
by Aymeric Moizard 08 Jan '20

08 Jan '20

Hi List, History: * In the past, I had deadlock which was, most probably, related to ssl1.1. We have discussed this issue, and a fix is supposed to workaround the issue that was detected. * With latest 5.2.X, I have experienced ONCE a similar behavior with TCP and TLS being mostly stuck. I have not been using this version much, but the fix was supposed to be in the core of kamailio. The status of the server this night: * I'm today running version: kamailio 5.3.1 (x86_64/linux), * Installed on stretch using http://deb.kamailio.org/kamailio53 repository. * This versions use libssl1.1 * A user reported that he can't connect with TCP * An average of 5000 IPs per 10 minutes are being banned by the pike module (could be twice the same) Yesterday/Today: * at the end of the outage, I had 2479 IP in my ipban htable. (which is equivalent to my statistics showing 2 bans/IP every 10 minutes = 5000) * looking at my logs, it appears that most (ALL?) ip being banned... are my regular users. * looking at my logs, I can't understand why pike would block them. This is a graph for statistics on my service for the last 24 hours: https://www.antisip.com/sip-antisip-com-register/status2.html Yesterday, at 22:18:39, kamailio started to BAN some IPs. 52 IPs were banned in a period of 10 minutes. I can confirm this from my logs. My pike configuration is this one: modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 64) modparam("pike", "remove_latency", 4) When detecting the issue, this morning, I typed: $> sudo kamctl stats $> sudo kamcmd htable.dump ipban //FAILURE (answer too large...) $> sudo kamctl trap Then, I started an agent with TCP and it worked...??? Then, a few seconds, may be a minute after: $> sudo kamcmd htable.dump ipban //SUCCESS and shows 2479 banned ip. and... everything is back to normal in a few minutes. I haven't restarted kamailio, and all statistics are as expected, as usual. Thus, it looks that " sudo kamctl trap" has triggered something. I already experienced a similar behavior -when testing my ssl1.1 deadlock last year-. 2 questions: 1/ I beleive my "pike" configuration should not ban users. Is my pike configuration wrong? As an example, pike has banned an IP sending one message/second. I believe my configuration should accept that? 2/ Could there still be a TLS issue with libssl1.1? This is the result of the "kamctl trap": https://sip.antisip.com/kamailio-pike-or-tls-issue-13-12-2019.kamctl-trap Sorry for the long story & hoping to find a long term solution or at least a workaround! Regards Aymeric -- Antisip - http://www.antisip.com

3 11

Re: [SR-Users] Add user issue - Multiicon
by technical＠multiicon.in 08 Jan '20

08 Jan '20

Dear Kamailio team, Greetings from Multiicon. We setuping kamailio sip server and all thing are done but when we going to add user we are getting Command not found. Using below command. I also check the server status by using systemctl status kamailio is showing active Please advise on this. Thanks & Regards, VIVEK AGHERA #technical department IND : +91 63548 23958 <mailto:vivek@multiicon.in> vivek(a)multiicon.in MULTIICON IDEOTECHNOLOGY PVT.LTD. 3rd Floor, Rumi Plaza, Airport Road, Rajkot - 360 001, Gujarat (INDIA) Please consider the environment before printing this e-mail.

2 1

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

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sr-users January 2020