sr-users November 2019

sr-users@lists.kamailio.org

67 participants
104 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

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Kamailio OPTIONS Round-Trip
by Soltanici Ilie 27 Mar '20

27 Mar '20

Hi, With Asterisk, we are able to get some peer round-trip connection statistic by setting qualify=yes for the specified peer. It sends periodic OPTIONS to the peer and calculates the time round trip time. It's something like - "Status: OK (30 ms)". Is there any way to achieve this in Kamailio by using nathelper module, or any other? Thank you.

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Kamailio stop to process incoming SIP traffic via TCP.
by Kristijan Vrban 16 Jan '20

16 Jan '20

Hi kamailios, i have a creepy situation with v5.2.1 stable Kamilio. After a day or so, Kamailio stop to process incoming SIP traffic via TCP. The incoming TCP network packages get TCP-ACK from the OS (Debian 9, 4.18.0-15-generic-Linux) but Kamailio does not show any processing for the SIP-Traffic incoming via TCP. No logs, nothing. While traffic via UDP is working just totally fine. When i look via command "netstat -ntp" is see, that the Recv-Q get bigger and bigger. e.g.: Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 4566 0 172.17.217.12:5060 xxx.xxx.xxx.xxx:57252 ESTABLISHED 31347/kamailio After Kamailio restart, all is working fine again for a day. We have maybe 10-20 devices online via TCP and low call volume (1-2 call per minute). The only settings for tcp we have is "tcp_delayed_ack=no" How to could we debug this situation? Again, no error, no warings in the log. Just nothing. Kristijan

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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.

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rtpproxy does not response, disable it
by Zodiac 03 Jan '20

03 Jan '20

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

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Not working : Using $cfg(line) to print line number
by mahesh b 24 Dec '19

24 Dec '19

Hi All, Below is the syntax am using to print the line number in kamailio configuration file. xlog("L_WARN","Hi Mahesh at line $cfg(line)\n"); Unfortunately , when i do this looks like kamailio is not starting, Any help please regarding what am i missing here. I tried Xlogl as well, which doesnt print the line number by default in my setup. This is the details i had read to understand config file attributes : $cfg(key) - Config File Attributes Attributes related to configuration file. The key can be: - line - return the current line in config - name - return the name of current config file - file - return the name of current config file - route - return the name of routing block Example: send_reply("404", "Not found at line $cfg(line)")

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lookup(aliases) issues with 5.2
by igor.potjevlesch＠gmail.com 13 Dec '19

13 Dec '19

Hello! I have notice some issues with lookup(aliases) since I upgraded to 5.2 from 5.1: First with kamctl: If this default command is executed: ctl_cmd_run ul.add "$USRLOC_TABLE" "$OSERUSER@$OSERDOMAIN" "$2" \ "$UL_EXPIRES" "$DEFAULT_Q" "$UL_PATH" "$UL_FLAGS" "$BR_FLAGS" "$ALL_METHODS" as $ALL_METHODS is not defined, there is a huge value in methods and then lookup(aliases) fails with -2. I have to update kamctl script and put -1 ("-1") instead of the variable. So, now, methods is a NULL value in DB and my alias is not ignored. But I use to make my alias like this: sip:number@blabla_1.local Which is no longer works: ERROR: registrar [common.c:62]: extract_aor(): failed to parse AoR [sip:number@blabla_1.local] Any idea of the difference since 5.2 with this behaviour? Without the underscore, the AoR seems to be valid but I need to support underscore as before. Regards, Igor.

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siputils: how does sip_p_charging_vector generate the header
by George Diamantopoulos 05 Dec '19

05 Dec '19

Hello, I've been looking at the sip_p_charging_vector() function of siputils. From the documentation it is not evident how the PCV header is generated. Are the exported $pcv pseudovariables writable, so that the function canl use those when generating the header? Thanks! BR, George

2 3

kamailio IMS supporting VOLTE
by Shubhendu Das 03 Dec '19

03 Dec '19

Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

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sr-users November 2019