sr-users April 2020

sr-users@lists.kamailio.org

70 participants
99 discussions

Problem with max tcp connections
by Jose Fco. Irles Durá 06 Apr '20

06 Apr '20

Hi I have a kamailio 5.1.2 as load balancer and registration offloading, but I have a problem with the max tcp connections that it can handle. I suspect that is a linux limit, but I don't find the reason or config. When that limit arrives, I can't connect to kamailio and I receive "Connection reset by peer", but I can't view any error message in the logs. If I check the connections in kamailio, I view that it have "free" connections: # kamctl kamcmd core.tcp_info { readers: 8 max_connections: 4096 max_tls_connections: 2048 opened_connections: 2655 opened_tls_connections: 0 write_queued_bytes: 0 } I have this configs in kamailio.conf (related to tcp) disable_tcp=no tcp_connection_lifetime=3610 tcp_connect_timeout=5 tcp_crlf_ping=yes tcp_accept_aliases=no tcp_keepalive=yes tcp_keepidle=5 tcp_rd_buf_size=65536 tcp_conn_wq_max=131072 mlock_pages=yes shm_force_alloc=yes tcp_max_connections=4096 The shm memory to 256 and the pkg memory to 32. And, following this doc: https://github.com/kamailio/kamailio/blob/master/doc/tutorials/tcp_tunning.… I have setted this values: net.ipv4.ip_local_port_range = 1024 65535 net.core.somaxconn = 65535 net.core.netdev_max_backlog = 182757 Also, I had checked the limits for the main process pid: Limit Soft Limit Hard Limit Units Max cpu time unlimited unlimited seconds Max file size unlimited unlimited bytes Max data size unlimited unlimited bytes Max stack size 8388608 unlimited bytes Max core file size unlimited unlimited bytes Max resident set unlimited unlimited bytes Max processes unlimited unlimited processes Max open files 1048576 1048576 files Max locked memory 16777216 16777216 bytes Max address space unlimited unlimited bytes Max file locks unlimited unlimited locks Max pending signals 386297 386297 signals Max msgqueue size 819200 819200 bytes Max nice priority 0 0 Max realtime priority 0 0 Max realtime timeout unlimited unlimited us The service is running inside a lxc container, without any resource limit, connected to the outside word throught macvlan interface. Where can I find problem source? Best regards

4 13

need help with RTPProxy in bridged mode
by Grant Bagdasarian 04 Apr '20

04 Apr '20

Hello, I'm using RTPproxy for the first time in bridged mode and I can't get kamailio/rtpproxy to rewrite the c parameter to the correct public ip address of kamailio. The setup is as following: Carrier ------[fiber]------ Kamailio ---------[lan]--------- Freeswitch Kamailio is listening on two interfaces: 1) Private: 172.0.0.1 2) Public: 192.168.0.1 (since we have a dedicated fiber with our carrier, this is its public address) Freeswitch is listening on: 1) 172.0.0.2 Carrier is on: 1) 10.0.0.1 I've started an rtpproxy instance on the Kamailio box using: rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.1 172.0.0.1 I've played around with rtpproxy_manage() and the various flags (ie, ei), but I can't get kamailio to set the correct public IP when the 200 OK has to be sent back to the carrier. It always sets it to its private address, instead of its public address. I'm using Kamailio 4.2 with sippy/rtpproxy 2.0. Could someone please point me into the right direction? Thanks! Grant

4 8

need help with RTPProxy in bridged mode
by Artur Hovsepyan 04 Apr '20

04 Apr '20

hi, i'm also having the same issue, same setup, but instead of private IP i have real IP on my WAN interface. and regardless that i've used rtpproxy_manage("coeir")and rtpproxy_manage("coier")kamailio is still replying with my internal (sip+rtp) ip in the outgoing to WAN SDP. Can you please help to understand how do i need to manager rtp proxy to send WAN IP in SDP? when my WAN IP is private IP and my carrier is also having private IP from same network, the issue is not seen. Thanks in advance,

1 1

rtpproxy bridge external(real ip)/internal
by Artur Hovsepyan 03 Apr '20

03 Apr '20

Hey, This is my first post here, so I wanted a bit of help. Appreciating that in advance. i have following setup: carrier(asterisk with sip ip=rtp ip=real IP) <==> kamailio+rtpproxy(ext real ip, internal ip) <==> internal asterisk(sip ip != rtp ip, both internal ips). rtprpoxy setup/usr/local/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy -F -s udp:127.0.0.1 7722 -l kam_int_ip kam_ext_ip -m 10000 -M 20000 -L 30000 -d DBUG LOG kamailio listens on both external and internal ipsdispatcher list: 0 towards carrier SOCKET: udp:external_ip2 towards internal asterisk SOCKET: udp:internal_ip calls being routed with if(src_ip==carrier's real ip) sent to route "from_carrier" and if(src_ip==int_ast_sip_ip) sent to route "to_carrier" in the routes calls are dispatched with if(!ds_select_dst("0", "4")) and if(!ds_select_dst("2", "4")) rtpproxy_manage is done in "NATMANAGE" route: #checking if call is from carrier or to carrier:if(src_ip!=carrier_ext_ip) { if(nat_uac_test("8")) { rtpproxy_manage("coier"); } else { rtpproxy_manage("cor"); }}else { rtpproxy_manage("coeir");} so, this setup works, when i setup private IP addresses(from same network) on both carrier side(10.0.1.1) and external interface of kamailio(10.0.1.2) (accordingly changing the dispatcher file, rtpproxy and the kamailio.cfg). as per below: carrier(asterisk with sip ip=rtp ip=10.0.1.1) <==> kamailio+rtpproxy(ext ip=10.0.1.2, internal ip=10.0.2.1) <==> internal asterisk(sip ip != rtp ip, both internal ips). rtprpoxy setup/usr/local/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy -F -s udp:127.0.0.1 7722 -l 10.0.2.1 10.0.1.2 -m 10000 -M 20000 -L 30000 -d DBUG LOG kamailio listens on both external and internal ipsdispatcher list: 0 towards carrier SOCKET: udp:10.0.1.22 towards internal asterisk SOCKET: udp:10.0.2.1 calls being routed with if(src_ip==10.0.1.1) sent to route "from_carrier" and if(src_ip==172.16.21.21) sent to route "to_carrier" in the routes calls are dispatched with if(!ds_select_dst("0", "4")) and if(!ds_select_dst("2", "4")) rtpproxy_manage is done in "NATMANAGE" route: #checking if call is from carrier or to carrier:if(src_ip!=10.0.1.1) { if(nat_uac_test("8")) { rtpproxy_manage("coier"); } else { rtpproxy_manage("cor"); }}else { rtpproxy_manage("coeir");} once i replace the IP addresses of carrier and kamailio with real IPs, kamailio is starting to forward internal asterisk's sip and rtp IP addresses to carrier and carrier's sip/rtp IPs in the SDP and breaking the SIP flow. that is is not the case while private IPs are setup for both external and internal interfaces. in the later case it normally does the bridging and sends correct info in SDP messages. can someone help me please to understand why/how it is working correctly for private IPs and "misbehaving" in case of real IPs. thanks,Art

1 1

Query on force_send_socket
by Amarnath K 03 Apr '20

03 Apr '20

Hi All, I'm using Kamailio version 5.1.9. In cfg file have routing to send out SIP messages from Kamailio stack. In this routing logic added force_send_socket("tcp:192.168.40.10:5060"); forward() I'm expecting the send port should be from 5060, but it always pick random port to send out message. my requirement is to send SIP messages from 5060 port instead of picking random port. Regards, Amarnath

5 6

Kamailio release
by Virgílio Cunha 03 Apr '20

03 Apr '20

Hi, When is the scheduled release date of kamailio-5.3.4? Best Regards, Virgílio Cunha

1 0

[ot] lightweight sip app for auto-answering with play file and echo mode
by Daniel-Constantin Mierla 01 Apr '20

01 Apr '20

Hello, wondering if anyone here is aware of a lightweight sip app that can answer a call, play some file and/or do echo mode, mainly targeted at using it for basic sip routing and call testing. Of course I know that Asterisk and FreeSwitch (or even SEMS) can do that, but they have many dependencies, requiring quite some resources to run them, so I thought maybe someone here figured out different solutions, eventually cli based apps like pjsua or baresip. GUI apps for Linux are also fine if they can be configured for such behaviour. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

7 18

Invite transaction lifetime
by Duarte Rocha 01 Apr '20

01 Apr '20

Greetings, I have a strange call with a client in which he sends me an INVITE and i do the usual "100", "183" and "200 - OK" but the client never gets my provisional or final responses and, as such keeps retransmitting the first INVITE. For the first retransmissions Kamailio doesn't relay the INVITE which i think is the correct behaviour. However, after around 30s, a retransmission is processed and is relayed to a different destination, using "ds_select_dst()" from the DISPATCH module. For this to happen it had to fail the "t_check_trans()" function from TM module which tells me that the transaction from the first INVITE was gone. I went to search timers and lifetime definitions on TM module and found one "max_inv_lifetime" which tells me that "An INVITE transaction will be kept in memory for maximum: max_inv_lifetime+fr_timer(from the ACK to the final reply wait)+wt_timer". In this case : max_inv_lifetime is 180 seconds, fr_timer is 30 seconds and wt_timer is 5 seconds. It's sum is 215seconds and from what i can see, my transaction only had a lifetime of 30s. fr_timer is configured with 30s but it's definition is "Timer which hits if no final reply for a request or ACK for a negative INVITE reply arrives (in milliseconds)." and it mentions ACK, but in my case it's an ACK for a final reply that is missing and not an ACK for a negative reply. What other reasons might have caused this transaction to be terminated ? Best Regards

1 0

Re: [SR-Users] Use topos module to hide contact in OPTIONS request
by Enzo 01 Apr '20

01 Apr '20

2 1

2026

2025

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 April 2020