Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-09 Thread lenz 

I have used this freeware tool in the past:  
http://sineapps.com/sinestatiax.php
maybe you can have a look at it as well
l.


In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd [EMAIL PROTECTED] ha  
scritto:

 At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
   At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
  
  How can I objectively measure jitter in Asterisk on a SIP channel?
  
  I don't just want to turn the new 1.4 jitter buffer on. I want to
  measure jitter.
  
  Thanks,
  Doug.

  You could look at the txjitter and rxjitter values (and other values)
  stored in the CHANNEL() function, and those values you're looking for
  were previously known as RTPAUDIOQOS.  Or is this not sufficient?

 Are txjitter and rxjitter working reliably? These calls are going to be
 placed from AMI and bridged together. Do you think the variables would
 be correctly set for each leg of the call?

 Doug.

 I think the best way to determine this would be to compare the
 numbers provided by CHANNEL() versus the numbers provided by
 something with a little more reliability, such as wireshark, in a
 controlled set of circumstances.

 Please post your results here - it would be an interesting test.

 JT

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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-09 Thread Douglas Garstang 
Oh jeez. Another GUI...

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of lenz
 Sent: Thursday, August 09, 2007 6:34 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
 
 
 I have used this freeware tool in the past:
 http://sineapps.com/sinestatiax.php
 maybe you can have a look at it as well
 l.
 
 
 In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd [EMAIL PROTECTED]
ha
 scritto:
 
  At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
   
   How can I objectively measure jitter in Asterisk on a SIP
channel?
   
   I don't just want to turn the new 1.4 jitter buffer on. I want
to
   measure jitter.
   
   Thanks,
   Doug.
 
   You could look at the txjitter and rxjitter values (and other
values)
   stored in the CHANNEL() function, and those values you're looking
for
   were previously known as RTPAUDIOQOS.  Or is this not sufficient?
 
  Are txjitter and rxjitter working reliably? These calls are going
to be
  placed from AMI and bridged together. Do you think the variables
would
  be correctly set for each leg of the call?
 
  Doug.
 
  I think the best way to determine this would be to compare the
  numbers provided by CHANNEL() versus the numbers provided by
  something with a little more reliability, such as wireshark, in a
  controlled set of circumstances.
 
  Please post your results here - it would be an interesting test.
 
  JT
 
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  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
 --
 Home of QueueMetrics - http://queuemetrics.com
 
 
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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-09 Thread Douglas Garstang 
 -Original Message-

 From: [EMAIL PROTECTED] [mailto:asterisk-users-

 [EMAIL PROTECTED] On Behalf Of John Todd

 Sent: Wednesday, August 08, 2007 5:08 PM

 To: Asterisk Users Mailing List - Non-Commercial Discussion

 Subject: Re: [asterisk-users] Measuring Jitter in Asterisk

 

 At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:

At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:

   

   How can I objectively measure jitter in Asterisk on a SIP
channel?

   

   I don't just want to turn the new 1.4 jitter buffer on. I want to

   measure jitter.

   

   Thanks,

   Doug.

 

   You could look at the txjitter and rxjitter values (and other
values)

   stored in the CHANNEL() function, and those values you're looking
for

   were previously known as RTPAUDIOQOS.  Or is this not sufficient?

 

 Are txjitter and rxjitter working reliably? These calls are going to
be

 placed from AMI and bridged together. Do you think the variables
would

 be correctly set for each leg of the call?

 

 Doug.

 

 I think the best way to determine this would be to compare the

 numbers provided by CHANNEL() versus the numbers provided by

 something with a little more reliability, such as wireshark, in a

 controlled set of circumstances.

 

 Please post your results here - it would be an interesting test.

 

No comparisons yet, but I may not need to.

I'm not feeling too confident with the figures in Asterisk to begin
with.

 

I had an Asterisk box, bridging two channels, where the media was going
to two different ITSP's. 

Upon hangup of the call, I was printing out the QoS stats available with
the CHANNEL(rtpqos) command. That seems to be what's implemented in
Asterisk 1.4.8.

 

h = {

Noop(local_ssrc = ${CHANNEL(rtpqos,audio,local_ssrc)});

Noop(local_lostpackets  =
${CHANNEL(rtpqos,audio,local_lostpackets)});

Noop(local_jitter   =
${CHANNEL(rtpqos,audio,local_jitter)});

Noop(local_count= ${CHANNEL(rtpqos,audio,local_count)});

Noop(remote_ssrc= ${CHANNEL(rtpqos,audio,remote_ssrc)});

Noop(remote_lostpackets =
${CHANNEL(rtpqos,audio,remote_lostpackets)});

Noop(remote_jitter  =
${CHANNEL(rtpqos,audio,remote_jitter)});

Noop(remote_count   =
${CHANNEL(rtpqos,audio,remote_count)});

Noop(rtt= ${CHANNEL(rtpqos,audio,rtt)});

}

 

When the call is hung up, I only see the output from this once. I'd
never thought about it before, but when you hang up a call, where two
channels are bridged, the hangup extension only gets called once for the
call, not once for each channel. Correct?

 

So, my output looked like this...

 

Connected to Asterisk 1.4.8 currently running on a1 (pid = 30914)

Verbosity is at least 3

a1*CLI show channels

Channel  Location State   Application(Data)


SIP/edge1-09bad778   (None)   Up  Bridged
Call(SIP/edge1-09baf35

SIP/edge1-09baf358   [EMAIL PROTECTED] Up
Dial(SIP/edge1/13033372500|60|

2 active channels

1 active call

  == Spawn extension (Outbound, 13033372500, 2) exited non-zero on
SIP/edge1-09baf358'

-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/edge1-09baf358, local_ssrc
= 891055531) in new stack

-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/edge1-09baf358,
local_lostpackets  = 1215) in new stack

-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/edge1-09baf358, local_jitter
= 3) in new stack

-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/edge1-09baf358, local_count
= 1124) in new stack

-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/edge1-09baf358, remote_ssrc
= 59917798) in new stack

-- Executing [EMAIL PROTECTED]:6] NoOp(SIP/edge1-09baf358,
remote_lostpackets = 1) in new stack

-- Executing [EMAIL PROTECTED]:7] NoOp(SIP/edge1-09baf358,
remote_jitter  = 0) in new stack

-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/edge1-09baf358, remote_count
= 1123) in new stack

-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/edge1-09baf358, rtt
= 0) in new stack

 

So, what do the totals represent? We're getting stats for two channels
added together it seems. Is local_jitter local jitter on both channels?
If so, it's completely useless. We need to be able to see stats for EACH
CHANNEL, otherwise they mean nothing.

 

Also, rtt is always 0. Man... the internet is fast today. Also,
local_lostpackets looks bogus. It's always some huge number, larger than
local_count.

 

Don't know if it's relevant, but this Asterisk box sent the call to an
edge router, than would sent the call onto the ITSP, and then drop out
of the RTP path. This Asterisk box was in the media, but the edge router
was not.

 

Doug.

 

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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-09 Thread Douglas Garstang 
I also just plugged a NoOp(${CHANNEL}) in the output. It does not matter
WHICH channel hangs up the call. The ${CHANNEL} variable is always set
to the second, outgoing call leg.

What does this mean? Why is that the case?

 

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, August 09, 2007 12:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Measuring Jitter in Asterisk

 

 -Original Message-

 From: [EMAIL PROTECTED] [mailto:asterisk-users-

 [EMAIL PROTECTED] On Behalf Of John Todd

 Sent: Wednesday, August 08, 2007 5:08 PM

 To: Asterisk Users Mailing List - Non-Commercial Discussion

 Subject: Re: [asterisk-users] Measuring Jitter in Asterisk

 

 At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:

At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:

   

   How can I objectively measure jitter in Asterisk on a SIP
channel?

   

   I don't just want to turn the new 1.4 jitter buffer on. I want to

   measure jitter.

   

   Thanks,

   Doug.

 

   You could look at the txjitter and rxjitter values (and other
values)

   stored in the CHANNEL() function, and those values you're looking
for

   were previously known as RTPAUDIOQOS.  Or is this not sufficient?

 

 Are txjitter and rxjitter working reliably? These calls are going to
be

 placed from AMI and bridged together. Do you think the variables
would

 be correctly set for each leg of the call?

 

 Doug.

 

 I think the best way to determine this would be to compare the

 numbers provided by CHANNEL() versus the numbers provided by

 something with a little more reliability, such as wireshark, in a

 controlled set of circumstances.

 

 Please post your results here - it would be an interesting test.

 

No comparisons yet, but I may not need to.

I'm not feeling too confident with the figures in Asterisk to begin
with.

 

I had an Asterisk box, bridging two channels, where the media was going
to two different ITSP's. 

Upon hangup of the call, I was printing out the QoS stats available with
the CHANNEL(rtpqos) command. That seems to be what's implemented in
Asterisk 1.4.8.

 

h = {

Noop(local_ssrc = ${CHANNEL(rtpqos,audio,local_ssrc)});

Noop(local_lostpackets  =
${CHANNEL(rtpqos,audio,local_lostpackets)});

Noop(local_jitter   =
${CHANNEL(rtpqos,audio,local_jitter)});

Noop(local_count= ${CHANNEL(rtpqos,audio,local_count)});

Noop(remote_ssrc= ${CHANNEL(rtpqos,audio,remote_ssrc)});

Noop(remote_lostpackets =
${CHANNEL(rtpqos,audio,remote_lostpackets)});

Noop(remote_jitter  =
${CHANNEL(rtpqos,audio,remote_jitter)});

Noop(remote_count   =
${CHANNEL(rtpqos,audio,remote_count)});

Noop(rtt= ${CHANNEL(rtpqos,audio,rtt)});

}

 

When the call is hung up, I only see the output from this once. I'd
never thought about it before, but when you hang up a call, where two
channels are bridged, the hangup extension only gets called once for the
call, not once for each channel. Correct?

 

So, my output looked like this...

 

Connected to Asterisk 1.4.8 currently running on a1 (pid = 30914)

Verbosity is at least 3

a1*CLI show channels

Channel  Location State   Application(Data)


SIP/edge1-09bad778   (None)   Up  Bridged
Call(SIP/edge1-09baf35

SIP/edge1-09baf358   [EMAIL PROTECTED] Up
Dial(SIP/edge1/13033372500|60|

2 active channels

1 active call

  == Spawn extension (Outbound, 13033372500, 2) exited non-zero on
SIP/edge1-09baf358'

-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/edge1-09baf358, local_ssrc
= 891055531) in new stack

-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/edge1-09baf358,
local_lostpackets  = 1215) in new stack

-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/edge1-09baf358, local_jitter
= 3) in new stack

-- Executing [EMAIL PROTECTED]:4] NoOp(SIP/edge1-09baf358, local_count
= 1124) in new stack

-- Executing [EMAIL PROTECTED]:5] NoOp(SIP/edge1-09baf358, remote_ssrc
= 59917798) in new stack

-- Executing [EMAIL PROTECTED]:6] NoOp(SIP/edge1-09baf358,
remote_lostpackets = 1) in new stack

-- Executing [EMAIL PROTECTED]:7] NoOp(SIP/edge1-09baf358,
remote_jitter  = 0) in new stack

-- Executing [EMAIL PROTECTED]:8] NoOp(SIP/edge1-09baf358, remote_count
= 1123) in new stack

-- Executing [EMAIL PROTECTED]:9] NoOp(SIP/edge1-09baf358, rtt
= 0) in new stack

 

So, what do the totals represent? We're getting stats for two channels
added together it seems. Is local_jitter local jitter on both channels?
If so, it's completely useless. We need to be able to see stats for EACH
CHANNEL, otherwise they mean nothing.

 

Also, rtt is always 0. Man... the internet is fast today. Also,
local_lostpackets looks bogus. It's always some huge number, larger than
local_count.

 

Don't know if it's

Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-08 Thread John Todd 
At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote:
   At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
  
  How can I objectively measure jitter in Asterisk on a SIP channel?
  
  I don't just want to turn the new 1.4 jitter buffer on. I want to
  measure jitter.
  
  Thanks,
  Doug.

  You could look at the txjitter and rxjitter values (and other values)
  stored in the CHANNEL() function, and those values you're looking for
  were previously known as RTPAUDIOQOS.  Or is this not sufficient?

Are txjitter and rxjitter working reliably? These calls are going to be
placed from AMI and bridged together. Do you think the variables would
be correctly set for each leg of the call?

Doug.

I think the best way to determine this would be to compare the 
numbers provided by CHANNEL() versus the numbers provided by 
something with a little more reliability, such as wireshark, in a 
controlled set of circumstances.

Please post your results here - it would be an interesting test.

JT

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[asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Douglas Garstang 
How can I objectively measure jitter in Asterisk on a SIP channel?

 

I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.

 

Thanks,

Doug.

 

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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Douglas Garstang 
Did a little research.

 

Wireshark can graph jitter measurement. That's cool, but pretty useless.

 

Now, what would be REALLY cool, was if tshark, the command line tool,
could measure jitter. It looks like it lacks this feature.

 

If it COULD, you could leave a tshark process running, constantly
measuring jitter in real time. You'd run one for each ITSP you use, and
voila, you have real time jitter metrics on a provider by provider
basis.

 

But... tshark doesn't' support this. Arrgh!

 

Doug.

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, August 03, 2007 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Measuring Jitter in Asterisk

 

How can I objectively measure jitter in Asterisk on a SIP channel?

 

I don't just want to turn the new 1.4 jitter buffer on. I want to
measure jitter.

 

Thanks,

Doug.

 

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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Alex Balashov 
On Fri, 3 Aug 2007, Douglas Garstang wrote:

 If it COULD, you could leave a tshark process running, constantly 
 measuring jitter in real time. You'd run one for each ITSP you use, and 
 voila, you have real time jitter metrics on a provider by provider 
 basis.

   There are various command-line SIP performance test tools (sipp?) that
can do this too, I think.

   Also, it may be possible to modify Wireshark's plugin to periodically
invoke its jitter analysis function automatically and export the results
to some retrievable location.  The most difficult problem would be
giving it a particular data stream to home in on as a VoIP call;  the
easiest thing there would be to nail up your own periodic tests from
a SIP UAC with definable IP endpoint locations and constantly run it
with that filter.

   Hackjobs aside, this sort of thing is essentially what products like
Brix do, as well as check in with SRTP stats.

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Jared Smith 
On Fri, 2007-08-03 at 12:31 -0700, Douglas Garstang wrote:
 How can I objectively measure jitter in Asterisk on a SIP channel?

 I don’t just want to turn the new 1.4 jitter buffer on. I want to
 measure jitter.

You can use Wireshark (formerly Ethereal) to analyze the RTP stream
after it's been captured.  You can either use Wireshark itself to do the
network capture, or you can capture the traffic with tcpdump and then
pull the file into Wireshark at a later time.

Inside Wireshark, go to Statistics, RTP, Show All Streams, and then
select a stream and hit the Analyze button.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Douglas Garstang 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Alex Balashov
 Sent: Friday, August 03, 2007 1:37 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
 
 On Fri, 3 Aug 2007, Douglas Garstang wrote:
 
  If it COULD, you could leave a tshark process running, constantly
  measuring jitter in real time. You'd run one for each ITSP you use,
and
  voila, you have real time jitter metrics on a provider by provider
  basis.
 
There are various command-line SIP performance test tools (sipp?)
that
 can do this too, I think.

I don't think you could do this with SIPP 

 
Also, it may be possible to modify Wireshark's plugin to
periodically
 invoke its jitter analysis function automatically and export the
results
 to some retrievable location.  The most difficult problem would be
 giving it a particular data stream to home in on as a VoIP call;  the
 easiest thing there would be to nail up your own periodic tests from
 a SIP UAC with definable IP endpoint locations and constantly run it
 with that filter.
 
Hackjobs aside, this sort of thing is essentially what products
like
 Brix do, as well as check in with SRTP stats.

Ok, maybe I should call them. But, as I said, if all their product does
is measure QoS and then give you pretty graphs to eyeball, it isn't much
use.

I need something that can measure jitter, latency etc in real time and
then stick the results somewhere, such as in MySQL. I can then choose
ITSP's based not just on route cost, but on a combination of route cost
and historical QoS data.

Doug.


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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Douglas Garstang 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jared Smith
 Sent: Friday, August 03, 2007 1:22 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
 
 On Fri, 2007-08-03 at 12:31 -0700, Douglas Garstang wrote:
  How can I objectively measure jitter in Asterisk on a SIP channel?
 
  I don't just want to turn the new 1.4 jitter buffer on. I want to
  measure jitter.
 
 You can use Wireshark (formerly Ethereal) to analyze the RTP stream
 after it's been captured.  You can either use Wireshark itself to do
the
 network capture, or you can capture the traffic with tcpdump and then
 pull the file into Wireshark at a later time.

Jared, that won't do. I don't want to run the wireshark GUI, and I don't
wan't to run it on every single Asterisk box, connecting back to a local
X server running on my desktop. I also don't want to capture the RTP
data, and store it somewhere for later analysis. I'm looking at a
situation here with millions of subscribers and dozens of ITSP's.

What I do want to do is record QoS data to every single ITSP in real
time. I can then lease cost route based not just on route cost, but also
on historical QoS data. Whatever tool is used to collect the QoS data
has to stick it somewhere, such as MySQL, and then when I route a call,
I will have to query that data from MySQL.

 
 Inside Wireshark, go to Statistics, RTP, Show All Streams, and then
 select a stream and hit the Analyze button.

I'm trying to avoid post-eyeballing the data.


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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Jared Smith 
On Fri, 2007-08-03 at 13:38 -0700, Douglas Garstang wrote:
 What I do want to do is record QoS data to every single ITSP in real
 time. 

If the provider sends RTCP packets, you could simply watch for those and
write the data to a database.  (I think modern versions of Asterisk even
allow you to get to the data from the dialplan, and possibly from the
Manager Interface.)  That at least gives you some per-call statistics.  

Beyond that, there's not a whole lot you can do with Asterisk itself,
unless someone gets around to writing a Call Quality Detail Record
module for Asterisk that would log the call quality stats on a
call-by-call basis.

Another option might be Packet Island's VoIPCare for Asterisk[1].  It
sounds like a nice solution, but I haven't tried it, so I can't say
whether or not it would work for you in your particular circumstances.

[1] http://www.packetisland.com/page-voipcare-for-asterisk.html


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Alex Balashov 
On Fri, 3 Aug 2007, Jared Smith wrote:

 If the provider sends RTCP packets, you could simply watch for those and
 write the data to a database.  (I think modern versions of Asterisk even
 allow you to get to the data from the dialplan, and possibly from the
 Manager Interface.)  That at least gives you some per-call statistics.

   If you want to go that route, just yank those packets out of a 
constantly running tcpdump process with the right filters, and then
process them with a script and load that data into a DB.

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Douglas Garstang 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Alex Balashov
 Sent: Friday, August 03, 2007 2:12 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
 
 On Fri, 3 Aug 2007, Jared Smith wrote:
 
  If the provider sends RTCP packets, you could simply watch for those
and
  write the data to a database.  (I think modern versions of Asterisk
even
  allow you to get to the data from the dialplan, and possibly from
the
  Manager Interface.)  That at least gives you some per-call
statistics.
 
If you want to go that route, just yank those packets out of a
 constantly running tcpdump process with the right filters, and then
 process them with a script and load that data into a DB.

Alex, ok... so if I wanted to measure jitter to an ITSP I could run
tcpdump to it, and parse the output. According to
http://wiki.wireshark.org/RTP_statistics, I'd have to compare the
timestamp in each RTP packet with the timestamp shown by tcpdump. Looks
kinda complicated.


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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread John Todd 
At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:

How can I objectively measure jitter in Asterisk on a SIP channel?

I don't just want to turn the new 1.4 jitter buffer on. I want to 
measure jitter.

Thanks,
Doug.

You could look at the txjitter and rxjitter values (and other values) 
stored in the CHANNEL() function, and those values you're looking for 
were previously known as RTPAUDIOQOS.  Or is this not sufficient?

I opened a request ticket to allow viewing of arbitrary CHANNEL() 
data on any active channel, but to my knowledge it has not been 
implemented.  The RTP source of media has however been impelemented 
in the CHANNEL() structure.  It may be possible to use chan_local to 
ascertain media data on the other leg of a call, but I have not 
experimented with that.

http://bugs.digium.com/view.php?id=9620

JT

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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Douglas Garstang 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of John Todd
 Sent: Friday, August 03, 2007 2:12 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
 
 At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
 
 How can I objectively measure jitter in Asterisk on a SIP channel?
 
 I don't just want to turn the new 1.4 jitter buffer on. I want to
 measure jitter.
 
 Thanks,
 Doug.
 
 You could look at the txjitter and rxjitter values (and other values)
 stored in the CHANNEL() function, and those values you're looking for
 were previously known as RTPAUDIOQOS.  Or is this not sufficient?

Thanks John. Missed those... they're not documented... not even in 'show
function CHANNEL'.

Doug.

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Re: [asterisk-users] Measuring Jitter in Asterisk

2007-08-03 Thread Douglas Garstang 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of John Todd
 Sent: Friday, August 03, 2007 2:12 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Measuring Jitter in Asterisk
 
 At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote:
 
 How can I objectively measure jitter in Asterisk on a SIP channel?
 
 I don't just want to turn the new 1.4 jitter buffer on. I want to
 measure jitter.
 
 Thanks,
 Doug.
 
 You could look at the txjitter and rxjitter values (and other values)
 stored in the CHANNEL() function, and those values you're looking for
 were previously known as RTPAUDIOQOS.  Or is this not sufficient?

Are txjitter and rxjitter working reliably? These calls are going to be
placed from AMI and bridged together. Do you think the variables would
be correctly set for each leg of the call?

Doug.

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