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dave at 3c.co.uk
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PostPosted: Mon Oct 06, 2008 9:28 pm    Post subject: [Freeswitch-users] Load test - performance not even matching Reply with quote

Going back a step, to where Jon was seeing more packets than thereshould have been, I've just encountered a similar issue having upgraded
to the latest, from what was probably a fairly old release - months old,
rather than weeks.


I've got two FS boxes (let's call them FS1 and FS2), each of which are
plumbed in to carrier C. There's an IVR service running on FS1; FS2
bridges any calls which it gets for said IVR over to FS1. What I've just had
is:
- calls from C to FS1 directly work fine;
- calls from C to FS2, thence to FS1 were silent. Looking at a capture from
FS2, everything looks OK except the RTP between FS1 and FS2. On answer,
there's a prompt played. What I see is three packets in a lump from FS1, then
four packets sent back from FS2 to FS1, four packets in a lump from FS1, then
five going back from FS2 to FS1, and so on.


The lumps are 20ms apart (codec is G711 with 20ms packets) - what seems to be
happening is that FS2 sends FS1 back the packets received from it unchanged
plus an extra packet which has arrived from C in the meantime.


FS2 ought to be sending these packets to C instead; it sends C nothing.


I've made the problem go away by commenting out the bit in switch_rtp.c which
auto-adjusts addresses (around line 1280.)


All of the machines have public IPs; there's not a NAT in sight.


I'll have a further look in the morning.


--Dave


Quote:
%(60000,0,300) means to generate a 60 second long 300hz tone
%(5,0,300) means a 5 ms long 300hz tone

if you are just trying to send a tone you are better off with
<action application="gentones" data="%(1000,0,300)|60"/>

which only generates 1 second of audio then buffers and loops it via the application
rather than allocating enough room for 60 seconds of signed linear audio and generating
the whole 60 seconds into memory for no reason vs 1 second sample looped 60 times.

No matter what you do it will not effect the bandwidth used, it's a factor of what codec you are using.


On Mon, Oct 6, 2008 at 4:15 AM, Jon Bruel <jbr@consiglia.dk (jbr@consiglia.dk)> wrote:
Quote:
Resolved: I have made further tests, and my final conclusion is that the
previous stated test results were screwed by the application 'gentones'.
This application does in some cases send more rtp than expected. If I
used:
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(60000,0,300)"/>
the expected rtp of 8600 kB/s was transmitted. If I used
<action application="gentones" data="%(60000,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>.
the rtp was 34600 kB/s, and the memory is heavily consumed. The only
difference being the sequence of the gentones commands. I don't know if
this is the expected behaviour of 'gentones' or not, but it certainly
screwed up the results previously posted. /Jon



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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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brian at freeswitch.org
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PostPosted: Mon Oct 06, 2008 9:35 pm    Post subject: [Freeswitch-users] Load test - performance not even matching Reply with quote

one tip is don't use -rtp_echo it sends 10 times the traffic back the other direction.

/b

On Oct 6, 2008, at 9:25 PM, David Knell wrote:
Quote:
Going back a step, to where Jon was seeing more packets than thereshould have been, I've just encountered a similar issue having upgraded
to the latest, from what was probably a fairly old release - months old,
rather than weeks.


I've got two FS boxes (let's call them FS1 and FS2), each of which are
plumbed in to carrier C. There's an IVR service running on FS1; FS2
bridges any calls which it gets for said IVR over to FS1. What I've just had
is:
- calls from C to FS1 directly work fine;
- calls from C to FS2, thence to FS1 were silent. Looking at a capture from
FS2, everything looks OK except the RTP between FS1 and FS2. On answer,
there's a prompt played. What I see is three packets in a lump from FS1, then
four packets sent back from FS2 to FS1, four packets in a lump from FS1, then
five going back from FS2 to FS1, and so on.


The lumps are 20ms apart (codec is G711 with 20ms packets) - what seems to be
happening is that FS2 sends FS1 back the packets received from it unchanged
plus an extra packet which has arrived from C in the meantime.


FS2 ought to be sending these packets to C instead; it sends C nothing.


I've made the problem go away by commenting out the bit in switch_rtp.c which
auto-adjusts addresses (around line 1280.)


All of the machines have public IPs; there's not a NAT in sight.


I'll have a further look in the morning.


--Dave


Quote:
%(60000,0,300) means to generate a 60 second long 300hz tone
%(5,0,300) means a 5 ms long 300hz tone

if you are just trying to send a tone you are better off with
<action application="gentones" data="%(1000,0,300)|60"/>

which only generates 1 second of audio then buffers and loops it via the application
rather than allocating enough room for 60 seconds of signed linear audio and generating
the whole 60 seconds into memory for no reason vs 1 second sample looped 60 times.

No matter what you do it will not effect the bandwidth used, it's a factor of what codec you are using.


On Mon, Oct 6, 2008 at 4:15 AM, Jon Bruel <jbr@consiglia.dk (jbr@consiglia.dk)> wrote:
Quote:
Resolved: I have made further tests, and my final conclusion is that the
previous stated test results were screwed by the application 'gentones'.
This application does in some cases send more rtp than expected. If I
used:
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(60000,0,300)"/>
the expected rtp of 8600 kB/s was transmitted. If I used
<action application="gentones" data="%(60000,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>.
the rtp was 34600 kB/s, and the memory is heavily consumed. The only
difference being the sequence of the gentones commands. I don't know if
this is the expected behaviour of 'gentones' or not, but it certainly
screwed up the results previously posted. /Jon



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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

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anthony.minessale at g...
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PostPosted: Tue Oct 07, 2008 8:41 am    Post subject: [Freeswitch-users] Load test - performance not even matching Reply with quote

are you doing anything with bypass_media?
We had another guy with something similar and we have not been able to nail it down besides
he was doing a media call that shifted to bypass_media mode.

If you have a situation where it happens every time a pcap and or FS console on debug of it happening would be nice.
It seems odd that the auto adjust code is being triggered considering the design says that if the leg from FS1 -> C gets a packet
from some other ip than that of C that it adjusts to that. So it would appear that somehow one of the packets from FS2 to FS1
ends up being picked up by the rtp socket that was meant for the FS1->C leg. (maybe a race in the port allocator giving both legs the same rtp port or something funny?)

say your phone is A

A -> FS1 (recv port 1000)
FS1->FS2 (recv port 1002)
FS2->C (recv port 1004)

either port 1004 is somehow getting the packet meant for 1002 which would make no sense
or somewhere in here both boxes picked the same port or the packet is somehow multicasting to both ports which would make no sense either.

I would like to figure it out though if you could provide the details including the dp and all the attrs you set etc.


BTW, you can turn off RTP auto ADJ both with a channel varable and a profile param (i forgot the name but it's on the wiki)









On Mon, Oct 6, 2008 at 9:25 PM, David Knell <dave@3c.co.uk (dave@3c.co.uk)> wrote:
Quote:
Going back a step, to where Jon was seeing more packets than thereshould have been, I've just encountered a similar issue having upgraded
to the latest, from what was probably a fairly old release - months old,
rather than weeks.


I've got two FS boxes (let's call them FS1 and FS2), each of which are
plumbed in to carrier C. There's an IVR service running on FS1; FS2
bridges any calls which it gets for said IVR over to FS1. What I've just had
is:
- calls from C to FS1 directly work fine;
- calls from C to FS2, thence to FS1 were silent. Looking at a capture from
FS2, everything looks OK except the RTP between FS1 and FS2. On answer,
there's a prompt played. What I see is three packets in a lump from FS1, then
four packets sent back from FS2 to FS1, four packets in a lump from FS1, then
five going back from FS2 to FS1, and so on.


The lumps are 20ms apart (codec is G711 with 20ms packets) - what seems to be
happening is that FS2 sends FS1 back the packets received from it unchanged
plus an extra packet which has arrived from C in the meantime.


FS2 ought to be sending these packets to C instead; it sends C nothing.


I've made the problem go away by commenting out the bit in switch_rtp.c which
auto-adjusts addresses (around line 1280.)


All of the machines have public IPs; there's not a NAT in sight.


I'll have a further look in the morning.


--Dave



Quote:
%(60000,0,300) means to generate a 60 second long 300hz tone
%(5,0,300) means a 5 ms long 300hz tone

if you are just trying to send a tone you are better off with
<action application="gentones" data="%(1000,0,300)|60"/>

which only generates 1 second of audio then buffers and loops it via the application
rather than allocating enough room for 60 seconds of signed linear audio and generating
the whole 60 seconds into memory for no reason vs 1 second sample looped 60 times.

No matter what you do it will not effect the bandwidth used, it's a factor of what codec you are using.


On Mon, Oct 6, 2008 at 4:15 AM, Jon Bruel <jbr@consiglia.dk (jbr@consiglia.dk)> wrote:
Quote:
Resolved: I have made further tests, and my final conclusion is that the
previous stated test results were screwed by the application 'gentones'.
This application does in some cases send more rtp than expected. If I
used:
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(60000,0,300)"/>
the expected rtp of 8600 kB/s was transmitted. If I used
<action application="gentones" data="%(60000,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>
<action application="gentones" data="%(5,0,300)"/>.
the rtp was 34600 kB/s, and the memory is heavily consumed. The only
difference being the sequence of the gentones commands. I don't know if
this is the expected behaviour of 'gentones' or not, but it certainly
screwed up the results previously posted. /Jon



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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
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jbr at consiglia.dk
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PostPosted: Thu Oct 09, 2008 9:49 am    Post subject: [Freeswitch-users] Load test - performance not even matching Reply with quote

I have been asked by Mark Crane and others to contribute more positively
by joining the community, and I would be glad to do so. As with anyone I
have my own agenda being to create a virtual PBX suite - using FS or
what ever I find most suitable. At this stage, I have added a load
testing page to the wiki, a modest first step to my contribution.
A few words about my findings regarding load tests: On the surface the
FS compares with the Asterisk: +/- 20% in terms of CPU load. The test
may not credit the FS for possible better resilience. Knowing Asterisk I
also know its instability under load, and with the architecture of the
FS, this may be resolved by design, and I'm looking forward to verifying
that. So I'll - in between the other tasks - continue with load tests
which attempt to simulate real life situations, with complex dial plans
and with erroneous internets. Ideas are welcome!
/Jon


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freeswitch-devel at no...
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PostPosted: Thu Oct 09, 2008 10:54 am    Post subject: [Freeswitch-users] Load test - performance not even matching Reply with quote

On Oct Thu 09 2008 16:36, Jon Bruel wrote:
Quote:
I have been asked by Mark Crane and others to contribute more positively
by joining the community, and I would be glad to do so. As with anyone I
have my own agenda being to create a virtual PBX suite - using FS or
what ever I find most suitable. At this stage, I have added a load
testing page to the wiki, a modest first step to my contribution.

Thanks a lot for your contribution Smile

Quote:
A few words about my findings regarding load tests: On the surface the
FS compares with the Asterisk: +/- 20% in terms of CPU load. The test
may not credit the FS for possible better resilience. Knowing Asterisk I
also know its instability under load, and with the architecture of the
FS, this may be resolved by design, and I'm looking forward to verifying
that. So I'll - in between the other tasks - continue with load tests
which attempt to simulate real life situations, with complex dial plans
and with erroneous internets. Ideas are welcome!

I am not a Load Test expert, but from an engineering point of view, I rather
first go for a "baseline suite" that would perform load testing under the
best conditions possible (that is, an average dial plan, and certainly not an
"erroneous internet" --I suppose you mean a lossy connection Confused).

This will give a reference result of the capabilities of FS (and any other
soft switch/pbx that can run the tests). Then, you can begin to push FS to
the limits, which will certainly test its resilience, architecture, etc.


If you go first for the real world, worst-case scenario, you might not get an
objective result. Imagine a test scenario such as the following: a web
browser that is incredibly fast rendering well-formed XML pages (such as
XHTML) but barks when rendering a malformed page. On the other hand, a crappy
browser, say with respect to speed or standards, but that has a best-effort
approach and manages to render, say 90% of a malformed page.

The first is a really good, speedy, browser, while the other is a crappy but
certainly resilient, best-effort, one. Broad, different results that have to
be viewed in their own frame of reference.


I would also try to approach the tests with an "atomic" point of view. That
is, if you go for a worst-case, real life, dense-in-features, scenario you
might find a multi-variable bug or whatever that might break FS and will be
hard to track because the test is itself multivariable, etc.

If you build tests by small, "atomic", steps, up to your real-life scenario,
where you to hit a nasty performance bug (or any other bug by this matter) it
will surely be much easier to track the reason triggering.


Finally, it would be wonderful to end up having the foundations for a load
test framework that would be able to test different switches / pbx
capabilities, much like the existing testing frameworks. Perhaps a nice
starting point would be

http://en.wikipedia.org/wiki/Load_testing
http://en.wikipedia.org/wiki/Performance_testing
http://en.wikipedia.org/wiki/Reliability_testing
http://en.wikipedia.org/wiki/Volume_testing

http://en.wikipedia.org/wiki/Category:Load_testing_tools


So these are my 2 cents Very Happy and again, take all this with a grain of salt
since IANALTE Cool

Cheers,

--
J. Javier Maestro <jjmaestro@nosys.es>
Socio Consultor - Nosys AJjV S.L.

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mike at jerris.com
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PostPosted: Thu Oct 09, 2008 5:58 pm    Post subject: [Freeswitch-users] Load test - performance not even matching Reply with quote

This is now fixed in trunk:

http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=9917

Mike

On Oct 6, 2008, at 10:25 PM, David Knell wrote:

Quote:
Going back a step, to where Jon was seeing more packets than there
should have been, I've just encountered a similar issue having
upgraded
to the latest, from what was probably a fairly old release - months
old,
rather than weeks.

I've got two FS boxes (let's call them FS1 and FS2), each of which are
plumbed in to carrier C. There's an IVR service running on FS1; FS2
bridges any calls which it gets for said IVR over to FS1. What I've
just had
is:
- calls from C to FS1 directly work fine;
- calls from C to FS2, thence to FS1 were silent. Looking at a
capture from
FS2, everything looks OK except the RTP between FS1 and FS2. On
answer,
there's a prompt played. What I see is three packets in a lump from
FS1, then
four packets sent back from FS2 to FS1, four packets in a lump from
FS1, then
five going back from FS2 to FS1, and so on.

The lumps are 20ms apart (codec is G711 with 20ms packets) - what
seems to be
happening is that FS2 sends FS1 back the packets received from it
unchanged
plus an extra packet which has arrived from C in the meantime.

FS2 ought to be sending these packets to C instead; it sends C
nothing.

I've made the problem go away by commenting out the bit in
switch_rtp.c which
auto-adjusts addresses (around line 1280.)

All of the machines have public IPs; there's not a NAT in sight.

I'll have a further look in the morning.

--Dave



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