Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or CANCEL. It'll just figure out what to do on its own.
None of this has to do with dialog state, though. Just rtpproxy control.
-- Alex
-- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz mino.haluz@gmail.com wrote:I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.
On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov lemenkov@gmail.com wrote:
2012/9/13 Mino Haluz mino.haluz@gmail.com:
Peter: Thanks for the tip! Really interesting. But I do not understand, why also this list contains the calls that were ended by sipp... Should I search for some mistake in my kamaillio config ?
Perhaps you don't close them with unforce_rtp_proxy:
if(method=="BYE" || method=="CANCEL"){ unforce_rtp_proxy(); }
-- With best regards, Peter Lemenkov.
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
_______________________________________________ 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
Ok, so I put there unforce_rtp_proxy even though I'm using rtpproxy_manage. The tip with nc now really shows the calls count.
But the dialog count is still higher and higher, so I have bug somewhere in the configuration. I'll check it.
On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov abalashov@evaristesys.com wrote:
Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or CANCEL. It'll just figure out what to do on its own.
None of this has to do with dialog state, though. Just rtpproxy control.
-- Alex
-- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/
Mino Haluz mino.haluz@gmail.com wrote: I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.
On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov lemenkov@gmail.com wrote:
2012/9/13 Mino Haluz mino.haluz@gmail.com:
Peter: Thanks for the tip! Really interesting. But I do not understand, why also this list contains the calls that were ended by sipp... Should I search for some mistake in my kamaillio config ?
Perhaps you don't close them with unforce_rtp_proxy:
if(method=="BYE" || method=="CANCEL"){ unforce_rtp_proxy(); }
-- With best regards, Peter Lemenkov.
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
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
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
The results:
- rtpproxy calls count 280 - sipp calls count 2000 - iptraf on proxy 4.8MB/s - G711a codec
So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), rtpproxy calls count is really the right value. CPU usage is ok on every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy cannot serve more than 270-280 calls ?
On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz mino.haluz@gmail.com wrote:
Ok, so I put there unforce_rtp_proxy even though I'm using rtpproxy_manage. The tip with nc now really shows the calls count.
But the dialog count is still higher and higher, so I have bug somewhere in the configuration. I'll check it.
On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov abalashov@evaristesys.com wrote:
Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or CANCEL. It'll just figure out what to do on its own.
None of this has to do with dialog state, though. Just rtpproxy control.
-- Alex
-- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard.
Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/
Mino Haluz mino.haluz@gmail.com wrote: I'm using rtpproxy_manage, so I assume unforce_rtp is not needed.
On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov lemenkov@gmail.com wrote:
2012/9/13 Mino Haluz mino.haluz@gmail.com:
Peter: Thanks for the tip! Really interesting. But I do not understand, why also this list contains the calls that were ended by sipp... Should I search for some mistake in my kamaillio config ?
Perhaps you don't close them with unforce_rtp_proxy:
if(method=="BYE" || method=="CANCEL"){ unforce_rtp_proxy(); }
-- With best regards, Peter Lemenkov.
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
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
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
Hello!
2012/9/13 Mino Haluz mino.haluz@gmail.com:
The results:
- rtpproxy calls count 280
- sipp calls count 2000
- iptraf on proxy 4.8MB/s
- G711a codec
So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), rtpproxy calls count is really the right value. CPU usage is ok on every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy cannot serve more than 270-280 calls ?
Let me be a dumb idiot here but did you set ulimits properly? 270-280 is pretty close to 256 (1024 / 4 ports).
You mean on the proxy side? I'm running rtpproxy as root, limits are still applied ? ulimit -s unlimited should do the trick ?
On Thu, Sep 13, 2012 at 6:25 PM, Peter Lemenkov lemenkov@gmail.com wrote:
Hello!
2012/9/13 Mino Haluz mino.haluz@gmail.com:
The results:
- rtpproxy calls count 280
- sipp calls count 2000
- iptraf on proxy 4.8MB/s
- G711a codec
So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), rtpproxy calls count is really the right value. CPU usage is ok on every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy cannot serve more than 270-280 calls ?
Let me be a dumb idiot here but did you set ulimits properly? 270-280 is pretty close to 256 (1024 / 4 ports). -- With best regards, Peter Lemenkov.
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
2012/9/13 Mino Haluz mino.haluz@gmail.com:
You mean on the proxy side? I'm running rtpproxy as root, limits are still applied ? ulimit -s unlimited should do the trick ?
Yes, they usually applied even for superuser, and yes - this should help (if that's the issue).
Thank you, setting ulimits worked! And the performance is the same as stated in the document I mentioned! :)
One more thing, I found these errors in syslog:
Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:211]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5270]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location
This is possibly something related to my sipp scenario, this is the INVITE sent from sipp
<![CDATA[
INVITE sip:800@perftest.vm SIP/2.0 Via: SIP/2.0/[transport] 10.0.2.36:[local_port];branch=[branch] From: "700" sip:500@10.0.2.36;tag=[call_number] To: sip:800@perftest.vm Call-ID: [call_id] CSeq: 1 INVITE Contact: "700" sip:700@10.0.2.36:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len]
v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] 10.0.2.36 s=- c=IN IP[local_ip_type] 10.0.2.36 t=0 0 m=audio [auto_media_port] RTP/AVP 8 a=rtpmap:8 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 ]]>
Do you see something that could cause this error ? Otherwise the call is initiated ok, but I really dont understand what is so strange to kamailio in this INVITE.
On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov lemenkov@gmail.com wrote:
2012/9/13 Mino Haluz mino.haluz@gmail.com:
You mean on the proxy side? I'm running rtpproxy as root, limits are still applied ? ulimit -s unlimited should do the trick ?
Yes, they usually applied even for superuser, and yes - this should help (if that's the issue).
-- With best regards, Peter Lemenkov.
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
Hello
On 09/13/2012 07:09 PM, Mino Haluz wrote:
Thank you, setting ulimits worked! And the performance is the same as stated in the document I mentioned! :)
One more thing, I found these errors in syslog:
Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:211]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5268]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location Sep 13 18:38:18 perftest kamailio[5270]: ERROR: <core> [parser/sdp/sdp.c:227]: Invalid payload location
This is possibly something related to my sipp scenario, this is the INVITE sent from sipp
<![CDATA[
INVITE sip:800@perftest.vm SIP/2.0 Via: SIP/2.0/[transport] 10.0.2.36:[local_port];branch=[branch] From: "700" <sip:500@10.0.2.36>;tag=[call_number] To: <sip:800@perftest.vm> Call-ID: [call_id] CSeq: 1 INVITE Contact: "700" <sip:700@10.0.2.36:[local_port]> Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len]v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] 10.0.2.36 s=- c=IN IP[local_ip_type] 10.0.2.36 t=0 0 m=audio [auto_media_port] RTP/AVP 8 a=rtpmap:8 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16
I think this SDP body is not correctly built. There are attributes for a media format 101 that is not declared in the m= line.
Regards
Javi
]]>
Do you see something that could cause this error ? Otherwise the call is initiated ok, but I really dont understand what is so strange to kamailio in this INVITE.
On Thu, Sep 13, 2012 at 6:37 PM, Peter Lemenkov lemenkov@gmail.com wrote:
2012/9/13 Mino Haluz mino.haluz@gmail.com:
You mean on the proxy side? I'm running rtpproxy as root, limits are still applied ? ulimit -s unlimited should do the trick ?
Yes, they usually applied even for superuser, and yes - this should help (if that's the issue).
-- With best regards, Peter Lemenkov.
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
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
-
Alex Balashov -
Javi Gallart -
Mino Haluz -
Peter Lemenkov