16 Aug
2016
16 Aug
'16
5:26 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
16 Aug
16 Aug
5:45 p.m.
On 16 Aug 2016, at 16:26, Jack Stevens
<Jack.Stevens(a)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(a)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>
5:47 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
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.stevens@netcall.com<mailto:jack.stevens@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-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(a)lists.sip-router.org>
Subject: Re: [SR-Users] Stress Testing
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
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
17 Aug
17 Aug
10:30 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
16 Aug
16 Aug
7:23 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(a)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
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(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
7:26 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(a)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
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
_______________________________________________
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
7:30 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(a)netcall.com>
To: Kamailio (SER) - Users Mailing List <sr-users(a)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(a)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
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
_______________________________________________
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
7:32 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(a)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(a)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
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(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3019
days inactive
3020
days old
7 comments
5 participants
participants (5)
-
Daniel Grotti -
Daniel-Constantin Mierla -
Dragos Oancea -
Jack Stevens -
Olle E. Johansson