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Core/ signal 11 Segmenation fault...

Kamailio SIP 500

Jack Stevens

16 Aug 2016 16 Aug '16

2:26 p.m.

Hi Guys,

I have been stress testing my Kamailio box but I am unable to get it upto 2000 concurrent calls it starts to fall over at 1300 have you got any ideas on how I can increase the performance of kamilio btw I am also using rtpengine

Kind Regards

-- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

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Olle E. Johansson

16 Aug 16 Aug

2:45 p.m.

...

On 16 Aug 2016, at 16:26, Jack Stevens Jack.Stevens@netcall.com wrote:

Hi Guys,

I have been stress testing my Kamailio box but I am unable to get it upto 2000 concurrent calls it starts to fall over at 1300 have you got any ideas on how I can increase the performance of kamilio btw I am also using rtpengine

Can you describe “fall over”

As Kamailio doesn’t handle media it’s likely an issue with rtpengine and the developers there needs to respond, but some more facts would be good. :-)

...

Kind Regards

CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED

This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation

Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Jack Stevens

2:47 p.m.

Hi,

Yeah we get calls time out please see the below

SIP messages for Call-ID bdc7441b322047868e294188984bbd09

15:46:49.599 [+0.00ms] [TX] INVITE to 192.168.3.204:5060 INVITE sip:12345@192.168.3.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Max-Forwards: 70 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204 Contact: sip:55123@192.168.1.79:5070 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE Supported: 100rel, timer User-Agent: StarTrinity.SIP 2016-07-13 16.10 UTC Session-Expires: 3600;refresher=uac Content-Type: application/sdp Content-Length: 329

v=0 o=- 3680351217 3680351217 IN IP4 192.168.1.79 s=o14160.proxy.stream0 c=IN IP4 192.168.1.79 t=0 0 m=audio 16114 RTP/AVP 8 0 4 18 101 a=rtcp:16115 IN IP4 192.168.1.79 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15

============================================================end of message================= 15:46:49.604 [+5.11ms] [RX] trying -- your call is important to us from 192.168.3.204:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.1.79:5070;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b;received=192.168.1.79 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Content-Length: 0

============================================================end of message================= 15:46:51.554 [+1,955.27ms] [RX] Request Timeout from 192.168.3.204:5060 SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 192.168.1.79:5070;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b;received=192.168.1.79 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Content-Length: 0

============================================================end of message================= 15:46:51.554 [+1,955.29ms] [TX] ACK to 192.168.3.204:5060 ACK sip:12345@192.168.3.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Max-Forwards: 70 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 ACK Content-Length: 0

============================================================end of message================= 15:46:59.312 [+9,713.29ms] [RX] Trying from 192.168.3.204:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Length: 0

============================================================end of message================= 15:46:59.312 [+9,713.43ms] [RX] OK from 192.168.3.204:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=as5772e8de Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Type: application/sdp Require: timer Content-Length: 258

v=0 o=root 1651160446 1651160446 IN IP4 10.200.0.57 s=Asterisk PBX 11.20.0 c=IN IP4 10.200.0.57 t=0 0 m=audio 10720 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

============================================================end of message================= 15:46:59.312 [+9,713.44ms] [TX] ACK to 192.168.3.204:5060 ACK sip:12345@192.168.3.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Max-Forwards: 70 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 ACK Content-Length: 0

============================================================end of message================= 15:46:59.313 [+9,714.41ms] [RX] Trying from 192.168.3.204:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Length: 0

============================================================end of message================= 15:46:59.314 [+9,714.49ms] [RX] OK from 192.168.3.204:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=as5772e8de Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Type: application/sdp Require: timer Content-Length: 258

v=0 o=root 1651160446 1651160447 IN IP4 10.200.0.57 s=Asterisk PBX 11.20.0 c=IN IP4 10.200.0.57 t=0 0 m=audio 10720 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

============================================================end of message================= 15:46:59.314 [+9,714.50ms] [TX] ACK to 192.168.3.204:5060 ACK sip:12345@192.168.3.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Max-Forwards: 70 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 ACK Content-Length: 0

============================================================end of message================= 15:46:59.450 [+9,850.82ms] [RX] Trying from 192.168.3.204:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Length: 0

============================================================end of message================= 15:46:59.450 [+9,850.84ms] [RX] OK from 192.168.3.204:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=as5772e8de Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Type: application/sdp Require: timer Content-Length: 258

v=0 o=root 1651160446 1651160448 IN IP4 10.200.0.57 s=Asterisk PBX 11.20.0 c=IN IP4 10.200.0.57 t=0 0 m=audio 10720 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv

============================================================end of message================= 15:46:59.450 [+9,850.86ms] [TX] ACK to 192.168.3.204:5060 ACK sip:12345@192.168.3.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Max-Forwards: 70 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 ACK Content-Length: 0

============================================================end of message================= 15:46:59.853 [+10,253.75ms] [RX] OK from 192.168.3.204:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=as5772e8de Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Type: application/sdp Require: timer Content-Length: 258

============================================================end of message================= 15:46:59.853 [+10,253.77ms] [TX] ACK to 192.168.3.204:5060 ACK sip:12345@192.168.3.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Max-Forwards: 70 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 ACK Content-Length: 0

============================================================end of message================= 15:47:00.800 [+11,200.72ms] [RX] OK from 192.168.3.204:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=as5772e8de Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Type: application/sdp Require: timer Content-Length: 258

============================================================end of message================= 15:47:00.800 [+11,200.73ms] [TX] ACK to 192.168.3.204:5060 ACK sip:12345@192.168.3.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Max-Forwards: 70 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 ACK Content-Length: 0

============================================================end of message================= 15:47:02.811 [+13,211.65ms] [RX] OK from 192.168.3.204:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.79:5070;received=192.168.1.79;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Record-Route: sip:10.200.0.56;line=sr-mUuGcDnGhDNG4FYGhD8UcPNa1TKecUya1TKe Record-Route: sip:192.168.3.204;r2=on;lr=on From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=as5772e8de Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 INVITE Server: FPBX-2.11.0(11.20.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uac Contact: sip:10.200.0.56;line=sr-mUuGcDna4b8g8qnGhDNG4FYGhDLPcDLGvDB* Content-Type: application/sdp Require: timer Content-Length: 258

============================================================end of message================= 15:47:02.811 [+13,211.67ms] [TX] ACK to 192.168.3.204:5060 ACK sip:12345@192.168.3.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b Max-Forwards: 70 From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007 Call-ID: bdc7441b322047868e294188984bbd09 CSeq: 5624 ACK Content-Length: 0

============================================================end of message================= ===============================saved by StarTrinity SIP Tester at 16/08/2016 15:47:05======

Jack Stevens

Cloud Systems and Network Administrator

Netcall

t 0330 333 6100

f 0330 333 0102

e jack.stevens@netcall.commailto:jack.stevens@netcall.com

w www.netcall.comhttp://www.netcall.com

b www.netcall.com/bloghttp://www.netcall.com/blog

n www.netcall.com/subscribehttp://www.netcall.com/subscribe

From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Olle E. Johansson Sent: 16 August 2016 15:45 To: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Subject: Re: [SR-Users] Stress Testing

On 16 Aug 2016, at 16:26, Jack Stevens <Jack.Stevens@netcall.commailto:Jack.Stevens@netcall.com> wrote:

Hi Guys,

As Kamailio doesn’t handle media it’s likely an issue with rtpengine and the developers there needs to respond, but some more facts would be good. :-)

Kind Regards

CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED

This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation

Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.orgmailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Daniel-Constantin Mierla

17 Aug 17 Aug

7:30 a.m.

Hello,

if the timeout is sent back by kamailio, then it tried to forward the request but the next hop didn't reply. The next hop may be over capacity or, if the next hop is not responding fast, be sure you have adequate values for tm timeout parameters.

Cheers, Daniel

On 16/08/16 16:47, Jack Stevens wrote:

...

Hi,

Yeah we get calls time out please see the below

SIP messages for Call-ID bdc7441b322047868e294188984bbd09

15:46:49.599 [+0.00ms] [TX] INVITE to 192.168.3.204:5060

INVITE sip:12345@192.168.3.204:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b

Max-Forwards: 70

From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d

To: sip:12345@192.168.3.204

Contact: sip:55123@192.168.1.79:5070

Call-ID: bdc7441b322047868e294188984bbd09

CSeq: 5624 INVITE

Allow: INFO, PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE

Supported: 100rel, timer

User-Agent: StarTrinity.SIP 2016-07-13 16.10 UTC

Session-Expires: 3600;refresher=uac

Content-Type: application/sdp

Content-Length: 329

v=0

o=- 3680351217 3680351217 IN IP4 192.168.1.79

s=o14160.proxy.stream0

c=IN IP4 192.168.1.79

t=0 0

m=audio 16114 RTP/AVP 8 0 4 18 101

a=rtcp:16115 IN IP4 192.168.1.79

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:4 G723/8000

a=rtpmap:18 G729/8000

a=sendrecv

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

============================================================end of message=================

15:46:49.604 [+5.11ms] [RX] trying -- your call is important to us from 192.168.3.204:5060

SIP/2.0 100 trying -- your call is important to us

Via: SIP/2.0/UDP 192.168.1.79:5070;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b;received=192.168.1.79

From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d

To: sip:12345@192.168.3.204

Call-ID: bdc7441b322047868e294188984bbd09

CSeq: 5624 INVITE

Content-Length: 0

============================================================end of message=================

15:46:51.554 [+1,955.27ms] [RX] Request Timeout from 192.168.3.204:5060

SIP/2.0 408 Request Timeout

Via: SIP/2.0/UDP 192.168.1.79:5070;rport=5070;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b;received=192.168.1.79

From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d

To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007

Call-ID: bdc7441b322047868e294188984bbd09

CSeq: 5624 INVITE

Content-Length: 0

============================================================end of message=================

15:46:51.554 [+1,955.29ms] [TX] ACK to 192.168.3.204:5060

ACK sip:12345@192.168.3.204:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.79:5070;rport;branch=z9hG4bKPj27345a1420fb41f89b76cce55b29282b

Max-Forwards: 70

From: sip:55123@192.168.3.204;tag=d8efe9d097c442478ede1f148cb2271d

To: sip:12345@192.168.3.204;tag=604fc74b9f0d3ee4c15ae560ab8f892f-8007

Call-ID: bdc7441b322047868e294188984bbd09

CSeq: 5624 ACK

Content-Length: 0

-- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

Daniel Grotti

16 Aug 16 Aug

4:23 p.m.

Hi, a couple of suggestions:

you may running out of RTP ports. kamailio may start to block requests if you are using pike module. UAC may start to de-register after a short expires time, so they won't be reachable anymore if they do not refresh their registration. You may check your natping interval, if you have it enable. It can cause some flood. you may check rtpengine TIMEOUT/SILENT_TIMEOUT. If you don't play RTP stream during the test, calls may be dropped after that timeout.

cheers, Daniel

On 08/16/2016 04:45 PM, Olle E. Johansson wrote:

...

...
On 16 Aug 2016, at 16:26, Jack Stevens <Jack.Stevens@netcall.com mailto:Jack.Stevens@netcall.com> wrote:

Hi Guys, I have been stress testing my Kamailio box but I am unable to get it upto 2000 concurrent calls it starts to fall over at 1300 have you got any ideas on how I can increase the performance of kamilio btw I am also using rtpengine

Can you describe “fall over”

As Kamailio doesn’t handle media it’s likely an issue with rtpengine and the developers there needs to respond, but some more facts would be good. :-)

/O

...
Kind Regards

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Jack Stevens

4:26 p.m.

Hi Dan,

We have checked all of that and we also not using pike its really strange

From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel Grotti Sent: 16 August 2016 17:24 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Stress Testing

Hi, a couple of suggestions:

cheers, Daniel

On 08/16/2016 04:45 PM, Olle E. Johansson wrote:

On 16 Aug 2016, at 16:26, Jack Stevens <Jack.Stevens@netcall.commailto:Jack.Stevens@netcall.com> wrote:

Hi Guys,

As Kamailio doesn’t handle media it’s likely an issue with rtpengine and the developers there needs to respond, but some more facts would be good. :-)

Kind Regards

CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED

_______________________________________________

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Daniel Grotti

4:30 p.m.

Then you should check kamailio.log and see why it start to fail. It should be quite easy to see why it start to fail from the log.

Daniel

----- Original Message ----- From: Jack Stevens Jack.Stevens@netcall.com To: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org Sent: Tue, 16 Aug 2016 18:26:55 +0200 (CEST) Subject: Re: [SR-Users] Stress Testing

Hi Dan,

We have checked all of that and we also not using pike its really strange

From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Daniel Grotti Sent: 16 August 2016 17:24 To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Stress Testing

Hi, a couple of suggestions:

cheers, Daniel

On 08/16/2016 04:45 PM, Olle E. Johansson wrote:

On 16 Aug 2016, at 16:26, Jack Stevens <Jack.Stevens@netcall.commailto:Jack.Stevens@netcall.com> wrote:

Hi Guys,

As Kamailio doesn’t handle media it’s likely an issue with rtpengine and the developers there needs to respond, but some more facts would be good. :-)

Kind Regards

CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED

_______________________________________________

SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list

sr-users@lists.sip-router.orgmailto:sr-users@lists.sip-router.org

http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

Dragos Oancea

4:32 p.m.

Hi,

Perhaps you could try adjusting your sysctl params:

Here are some values, but they should be adjusted for your machine: net.core.rmem_max=16777216 net.core.wmem_max=16777216 net.ipv4.tcp_rmem=4096 87380 16777216 net.ipv4.tcp_wmem=4096 65536 16777216 net.ipv4.udp_mem=16777216 16777216 16777216 net.ipv4.udp_rmem_min=16777216 net.ipv4.udp_wmem_min=16777216 net.core.netdev_max_backlog = 50000

kernel.shmall = 2097152 kernel.shmmax = 2147483648 kernel.shmmni = 4096 kernel.sem = 250 32000 100 128

Cheers, Dragos

On 16/08/2016 18:26, Jack Stevens wrote:

...

Hi Dan,

We have checked all of that and we also not using pike its really strange

*From:*sr-users [mailto:sr-users-bounces@lists.sip-router.org] *On Behalf Of *Daniel Grotti *Sent:* 16 August 2016 17:24 *To:* sr-users@lists.sip-router.org *Subject:* Re: [SR-Users] Stress Testing

Hi, a couple of suggestions:

you may running out of RTP ports. kamailio may start to block requests if you are using pike module. UAC may start to de-register after a short expires time, so they won't be reachable anymore if they do not refresh their registration. You may check your natping interval, if you have it enable. It can cause some flood. you may check rtpengine TIMEOUT/SILENT_TIMEOUT. If you don't play RTP stream during the test, calls may be dropped after that timeout.

cheers, Daniel

On 08/16/2016 04:45 PM, Olle E. Johansson wrote:
    On 16 Aug 2016, at 16:26, Jack Stevens <Jack.Stevens@netcall.com
    <mailto:Jack.Stevens@netcall.com>> wrote:



    Hi Guys,



    I have been stress testing my Kamailio box but I am unable to
    get it upto 2000 concurrent calls it starts to fall over at 1300
    have you got any ideas on how I can increase the performance of
    kamilio btw I am also using rtpengine

Can you describe “fall over”



As Kamailio doesn’t handle media it’s likely an issue with rtpengine
and the developers there needs to respond,

but some more facts would be good. :-)



/O



    Kind Regards







    CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED

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    addressee. It may contain private and/or confidential
    information. If you have received this email in error you are on
    notice of its status, please immediately notify the sender by
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    you are not the addressee, except to notify the sender, you must
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    attachments. Netcall Telecom accepts no responsibility for any
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    original author. Opinions or views expressed in this email may
    be those of the individual sender and not Netcall Telecom.
    Nothing in this email shall bind Netcall Telecom in any contract
    or obligation

    Netcall Telecom Ltd Registered in England 2831215. Registered
    Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel
    Hempstead, Herts, HP1 1BB
    _______________________________________________
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This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation

Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

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7 comments

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participants (5)

Daniel Grotti
Daniel-Constantin Mierla
Dragos Oancea
Jack Stevens
Olle E. Johansson