13 Sep
2012
13 Sep
'12
2:53 p.m.
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(a)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(a)gmail.com> wrote:
2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>om>:
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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.
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13 Sep
13 Sep
3:07 p.m.
New subject: RTPProxy with kamailio : How to get calls count?
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(a)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(a)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(a)gmail.com> wrote:
2012/9/13 Mino Haluz
<mino.haluz(a)gmail.com>om>:
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_______________________________________________
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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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3:52 p.m.
New subject: RTPProxy with kamailio : How to get calls count?
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(a)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(a)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(a)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(a)gmail.com> wrote:
>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>om>:
>>
>>> 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(a)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
>
> _______________________________________________
> 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
>
4:25 p.m.
New subject: RTPProxy with kamailio : How to get calls count?
Hello!
2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>om>:
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.
4:31 p.m.
New subject: RTPProxy with kamailio : How to get calls count?
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(a)gmail.com> wrote:
Hello!
2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>om>:
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4:37 p.m.
New subject: RTPProxy with kamailio : How to get calls count?
2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>om>:
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.
5:09 p.m.
New subject: RTPProxy with kamailio : How to get calls count?
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(a)gmail.com> wrote:
2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>om>:
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 
14 Sep
14 Sep
8:08 a.m.
New subject: RTPProxy with kamailio : How to get calls count?
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(a)gmail.com> wrote:
2012/9/13 Mino Haluz
<mino.haluz(a)gmail.com>om>:
_______________________________________________
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4323
days inactive
4324
days old
7 comments
4 participants
participants (4)
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Alex Balashov -
Javi Gallart -
Mino Haluz -
Peter Lemenkov