[asterisk-users] RTPAUDIOQOS - Depending on who hangs up the phone, it's empty
I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes I
have asterisk sending QOS data to the console. It seems I get QOS data only
if the caller hangs up, with the variable being empty if the callee (or
asterisk) hangs up.
Any idea why I would see this?
exten = h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS})
Thanks,
Richard Seguin
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Re: [asterisk-users] RTPAUDIOQOS
OK, how do you get such information -- at times it would be very useful to know. Darryl Dunkin ddun...@netos.net wrote: Sorry to reply so late, I am months behind and catching up. I have been inspecting this on my own systems, and the results are inconsistent to say the least. I’ve been dumping these to the verbose logs for some time and monitoring them, but I have not been able to determine why the numbers are so far off. I am more concerned with the packets lost due to priority queuing within our network. Here is an example just today: ssrc=583450581 themssrc=1093951555 lp=0 rxjitter=0.003219 rxcount=1100 txjitter=0.000275 txcount=1108 rlp=57702 rtt=0.036000 If the txcount is only 1108, how can the remote lost packet count be 57702? Unless the call was nearly inaudible? I did verify with this end user, and the call was just fine. Is this an issue with the phone at the remote end misreporting? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys Sent: Tuesday, September 22, 2009 01:01 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] RTPAUDIOQOS Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: 2009 m. rugsėjo 22 d. 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval Alternatives: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
I add this line in our in/out contexts:
exten = h,1,Noop(QOS=${RTPAUDIOQOS})
Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging on).
I'm sure you could output it anwhere else as well with a system call/echo.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Thursday, November 12, 2009 06:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTPAUDIOQOS
OK, how do you get such information -- at times it would be very useful
to know.
Darryl Dunkin ddun...@netos.net wrote:
Sorry to reply so late, I am months behind and catching up.
I have been inspecting this on my own systems, and the results are
inconsistent to say the least. I’ve been dumping these to the verbose logs
for some time and monitoring them, but I have not been able to determine why
the numbers are so far off. I am more concerned with the packets lost due to
priority queuing within our network.
Here is an example just today:
ssrc=583450581
themssrc=1093951555
lp=0
rxjitter=0.003219
rxcount=1100
txjitter=0.000275
txcount=1108
rlp=57702
rtt=0.036000
If the txcount is only 1108, how can the remote lost packet count be 57702?
Unless the call was nearly inaudible?
I did verify with this end user, and the call was just fine. Is this an issue
with the phone at the remote end misreporting?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys
Sent: Tuesday, September 22, 2009 01:01
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] RTPAUDIOQOS
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA
Sent: 2009 m. rugsėjo 22 d. 09:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RTPAUDIOQOS
hey all,
can any body know what this parameter stands for
i got RTPAUDIOQOS while i have SIP channels
but could not understand then what this parameter tell
ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000
if any one know plese help me to or give any documentation link
regards
Dhaval
Alternatives:
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How do
you spend it?
John Covici
cov...@ccs.covici.com
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Re: [asterisk-users] RTPAUDIOQOS
OK, thanks -- will have to try and see what I get.
Darryl Dunkin ddun...@netos.net wrote:
I add this line in our in/out contexts:
exten = h,1,Noop(QOS=${RTPAUDIOQOS})
Then grep for 'QOS' in asterisk-verbose (assuming you have verbose logging
on). I'm sure you could output it anwhere else as well with a system
call/echo.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
cov...@ccs.covici.com
Sent: Thursday, November 12, 2009 06:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTPAUDIOQOS
OK, how do you get such information -- at times it would be very useful
to know.
Darryl Dunkin ddun...@netos.net wrote:
Sorry to reply so late, I am months behind and catching up.
I have been inspecting this on my own systems, and the results are
inconsistent to say the least. I’ve been dumping these to the verbose logs
for some time and monitoring them, but I have not been able to determine
why the numbers are so far off. I am more concerned with the packets lost
due to priority queuing within our network.
Here is an example just today:
ssrc=583450581
themssrc=1093951555
lp=0
rxjitter=0.003219
rxcount=1100
txjitter=0.000275
txcount=1108
rlp=57702
rtt=0.036000
If the txcount is only 1108, how can the remote lost packet count be 57702?
Unless the call was nearly inaudible?
I did verify with this end user, and the call was just fine. Is this an
issue with the phone at the remote end misreporting?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas
Kezys
Sent: Tuesday, September 22, 2009 01:01
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] RTPAUDIOQOS
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: 2009 m. rugsėjo 22 d. 09:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] RTPAUDIOQOS
hey all,
can any body know what this parameter stands for
i got RTPAUDIOQOS while i have SIP channels
but could not understand then what this parameter tell
ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000
if any one know plese help me to or give any documentation link
regards
Dhaval
Alternatives:
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--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici
cov...@ccs.covici.com
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How do
you spend it?
John Covici
cov...@ccs.covici.com
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Re: [asterisk-users] RTPAUDIOQOS
Sorry to reply so late, I am months behind and catching up. I have been inspecting this on my own systems, and the results are inconsistent to say the least. I’ve been dumping these to the verbose logs for some time and monitoring them, but I have not been able to determine why the numbers are so far off. I am more concerned with the packets lost due to priority queuing within our network. Here is an example just today: ssrc=583450581 themssrc=1093951555 lp=0 rxjitter=0.003219 rxcount=1100 txjitter=0.000275 txcount=1108 rlp=57702 rtt=0.036000 If the txcount is only 1108, how can the remote lost packet count be 57702? Unless the call was nearly inaudible? I did verify with this end user, and the call was just fine. Is this an issue with the phone at the remote end misreporting? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys Sent: Tuesday, September 22, 2009 01:01 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] RTPAUDIOQOS Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: 2009 m. rugsėjo 22 d. 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
Thank you for answer. It was very informative, I put it in our wiki if you don't mind. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: 2009 m. rugsėjo 22 d. 20:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTPAUDIOQOS On Tuesday 22 September 2009 10:42:44 Johann Steinwendtner wrote: Mindaugas Kezys schrieb: Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified In the given example: *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txji tter=0.00;txcount=83;rlp=0;rtt=14818.715000* How do I interprete the jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind this value ? It's a ratio of out-of-order (jittered) to in-order packets, calculated progressively. Due to the progressive calculation, it's not exactly 3/147, in this case, but it's close enough to know that 3 packets were received out-of-order. The closer the value is to 0, the better. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTPAUDIOQOS
hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell * ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000 * if any one know plese help me to or give any documentation link regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: 2009 m. rugsėjo 22 d. 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitt er=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
Have you looked at voip-info.org? I'm going to guess at what these mean and I'll bet a coffee that someone will correct what I say wrong: Ssrc = local ip address Themsrc = remote (provider) ip address Rxjitter = QOS variance for received jitter buffers Rxcount = packets received Txjitter = transmitted variance Txcount = packets transmitted Rlp = ?? Rtt = ?? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Tuesday, September 22, 2009 1:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitt er=0.00;txcount=83;rlp=0;rtt=14818.715000 if any one know plese help me to or give any documentation link regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
Mindaugas Kezys schrieb: Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified In the given example: *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000* How do I interprete the jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind this value ? Thanks Regards Hans *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *DHAVAL INDRODIYA *Sent:* 2009 m. rugsėjo 22 d. 09:28 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] RTPAUDIOQOS hey all, can any body know what this parameter stands for i got RTPAUDIOQOS while i have SIP channels but could not understand then what this parameter tell *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txjitter=0.00;txcount=83;rlp=0;rtt=14818.715000* if any one know plese help me to or give any documentation link regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS
On Tuesday 22 September 2009 10:42:44 Johann Steinwendtner wrote: Mindaugas Kezys schrieb: Check this link: http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified In the given example: *ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txji tter=0.00;txcount=83;rlp=0;rtt=14818.715000* How do I interprete the jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind this value ? It's a ratio of out-of-order (jittered) to in-order packets, calculated progressively. Due to the progressive calculation, it's not exactly 3/147, in this case, but it's close enough to know that 3 packets were received out-of-order. The closer the value is to 0, the better. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTPAUDIOQOS On DAHDI is it possible
hello I would like to take value RTPAUDIOQOS channel variable on DAHDI / IAX Channel... Any Idea..!! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS On DAHDI is it possible
10 sep 2009 kl. 12.33 skrev DHAVAL INDRODIYA: hello I would like to take value RTPAUDIOQOS channel variable on DAHDI / IAX Channel... DAHDI doesn't use the Realtime Transport Protocol, RTP. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS On DAHDI is it possible
thanks for your reply, i want to know about meetme recording while i Use SIP as meetme user i got RTPAUDIOQOS value but while in DAHDI i cannot get this. but for DAHDI channel how can I get QOS because some times my recording quality is very BAD is there any way to improve it? regards Dhaval On Thu, Sep 10, 2009 at 4:27 PM, Olle E. Johansson o...@edvina.net wrote: 10 sep 2009 kl. 12.33 skrev DHAVAL INDRODIYA: hello I would like to take value RTPAUDIOQOS channel variable on DAHDI / IAX Channel... DAHDI doesn't use the Realtime Transport Protocol, RTP. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTPAUDIOQOS On DAHDI is it possible
Neither RTPAUDIOQOS or any of the TRP qos facilities works correctly on asterisk like channel(...,...), nothing you can trust. Just forget it. Olivier DHAVAL INDRODIYA a crit: thanks for your reply, i want to know about meetme recording while i Use SIP as meetme user i got RTPAUDIOQOS value but while in DAHDI i cannot get this. but for DAHDI channel how can I get QOS because some times my recording quality is very BAD is there any way to improve it? regards Dhaval On Thu, Sep 10, 2009 at 4:27 PM, Olle E. Johansson o...@edvina.net wrote: 10 sep 2009 kl. 12.33 skrev DHAVAL INDRODIYA: hello I would like to take value RTPAUDIOQOS channel variable on DAHDI / IAX Channel... DAHDI doesn't use the Realtime Transport Protocol, RTP. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
