sr-users January 2020

sr-users@lists.kamailio.org

80 participants
93 discussions

by Alex Balashov

Hi, I wanted to get clarity on a rather basic question: When using Kamailio with TCP, is it possible to end up with fragments of multiple SIP messages in a single "on-the-wire" packet? >From what I understand, the purpose of Kamailio's setting TCP_NODELAY is to prevent such bundling--as much as it is possible to control the fragmentation/packetisation behaviour of the transport protocol from a user-space application. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

4 years, 3 months

Huge DB load while using location and registrar module (save()).

by Voip support

Dear Community, I am facing an issue with using the usrloc and registrar module to save registration informations in database. It works for me but when tested yesterday with around 1500-1600 users i got a lot of queries on database mostly SELECT. I figured out that changing db_mode to 0 resolves the issue because it works totally in memory (almost no select query on DB). So i think its location module. I tried db_mode=3 and db_mode=2 but both create a lot of load on database. The only function i need is to store some information of REGISTER in DB to know the count of registered users and the source IP / user-agent / username data. It needs to be saved in table on successfull REGISTRATION and removed when registration is expired or somebody UN-REGISTER. When used location module for that there was lots of queries like this (20000-50000 per second): select `contact`,`expires`,`q`,`callid`,`cseq`,`flags`,`cflags`,`user_agent`,`received`,`path`,`socket`,`methods`,`last_modified`,`ruid`,`instance`,`reg_id`,`server_id`,`connection_id`,`keepalive` from `location` where `username`='xxxx' order by q update `location` set `expires`='2020-01-08 18:51:39',`q`=-1.000000 ,`cseq`=2042,`flags`=0,`cflags`=64,`user_agent`='n/a',`received`='sip:xxxx:5060',`path`=NULL,`socket`='udp:xxxx:20051',`methods`=NULL,`last_modified`='2020-01-08 18:50:39',`callid`='1205705227@xxxx ',`instance`=NULL,`reg_id`=0,`server_id`=0,`connection_id`=-1,`keepalive`=1,`contact`='sip:1019@xxx:5060' where `ruid`='uloc-5e1642a3-2b90-0161' Looking up for ideas how to resolve this issue. I was thinking of implementing INSERT UPDATE and DELETE based on SQLops. But how to recognize that REGISTER was expiered? I would like location module to work mostly like in-memory. For each new registration INSERT entry to DB, each REGISTER re-new an UPDATE or DELETE should be triggered. So i would expect for 1000 users around 2000 operations per minute. Best regards, Tom

4 years, 3 months

Kamailio v5.3.2 Released

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v5.3.2 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.2. For more details about version 5.3.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/01/kamailio-v5-3-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

4 years, 3 months

Ask for Siremis document

by Thanh Tran

Hi, I looking for Administration Document for Siremis but can't find it. Can you share document ? Thanks,

4 years, 3 months

Releasing Kamailio v5.3.2

by Daniel-Constantin Mierla

Hello, not so many fixes since the release of 5.3.1, but it looks enough to release a new minor version from branch 5.3. I am planning to do in on Thursday, Jan 9, 2020, if there are no new reports of relevant issues that make us to reconsider and postpone. Soon after we should probably do a release from branch 5.2 as well as the last one from branch 5.1. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

4 years, 3 months

Parsing exception when "user=phone" is present

by Duarte Rocha

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

4 years, 3 months

possible TCP deadlock (tls again?) // pike module not releasing IPs

by Aymeric Moizard

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

4 years, 3 months

Re: [SR-Users] Add user issue - Multiicon

by technical＠multiicon.in

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.

4 years, 3 months

WebRTC ACK Protocol

by Ilie Soltanici

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.

4 years, 3 months

Kamailio when acting as client doesnt send SNI in client hello handshake message

by mahesh b

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 years, 3 months

← Newer
1
2
3
4
5
6
7
8
9
10
Older →

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users January 2020