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 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================= 15:46:59.853 [+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 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================= 15:47:00.800 [+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 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================= 15:47:02.811 [+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.stev...@netcall.com<mailto:jack.stev...@netcall.com> w www.netcall.com<http://www.netcall.com> b www.netcall.com/blog<http://www.netcall.com/blog> n www.netcall.com/subscribe<http://www.netcall.com/subscribe> From: sr-users [mailto:sr-users-boun...@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.stev...@netcall.com<mailto:jack.stev...@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 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 -- 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 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
