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Explain output of show call active voice brief

jhun_puyong
Level 1
Level 1

Hi,

Could someone explain the output I have after I issue the command show call active voice brief.

Actually what I want to know is, what does noise:-52 acom:45  i/0:-51/-16 dBm and pl:146400/0ms lost:0/1/0 delay:55/55/85ms g711ulaw  TextRelay: off means?

Does the output tells the call is not answered or consider as a drop call?

Hope someone can explain me further. Thank you and appreciate the help.

816  : 53194 1107709670ms.1 +9960 pid:1 Answer 913103540000 active

dur 00:00:26 tx:7344/1175040 rx:7345/1233960

Tele 0/2/1:23 (53194) [0/2/1.5] tx:146870/146870/0ms g711ulaw noise:-52 acom:45  i/0:-51/-16 dBm

816  : 53195 1107709680ms.1 +9940 pid:3000 Originate 38562 active

dur 00:02:26 tx:7344/1175040 rx:7345/1175200

IP 10.204.21.212:25312 SRTP: off rtt:0ms pl:146400/0ms lost:0/1/0 delay:55/55/85ms g711ulaw TextRelay: off

media inactive detected:n media contrl rcvd:n/a timestamp:n/a

long duration call detected:n long duration call duration:n/a timestamp:n/a

3 Accepted Solutions

Accepted Solutions

Edson Pineiro
Level 1
Level 1

Hi Jhun,

To answer your question briefly the following two sections of the 'show call active voice brief' provide information regarding echo and QOS:

noise:-52 acom:45  i/0:-51/-16 dBm

-NoiseLevel=-59

The active noise level for this call.

This value is calculated in the comfort

noise generation module and is used

to generate comfort noise when voice

activity detection (VAD) is enabled.

-ACOMLevel=45

The current ACOM level for this call.

ACOM is the combined loss achieved

by the echo canceler. This value is the

sum of the Echo Return Loss (ERL),

Echo Return Loss Enhancement

(ERLE), and Non-Linear Processing

(NLP) loss for the call.

pl:146400/0ms lost:0/1/0 delay:55/55/85ms

-HiWaterPlayoutDelay=55ms

The First-In, First-Out (FIFO) jitter

buffer high mark that indicates the

maximum depth to which the de-jitter

buffer adapts for this call.

-LoWaterPlayoutDelay=55ms

The FIFO jitter buffer low mark that

indicates the minimum depth to which

the de-jitter buffer adapts for this call.

-ReceiveDelay=85 ms

The current playout FIFO delay plus

the decoder delay for the call.

The following output tells us that there was a codec negotiation using g711ulaw:

g711ulaw  TextRelay: off

Codec negotiation occurs after call setup and alerting states, so most likely the terminating device is off hook.

-be advised that the show call active voice brief only displays active calls, so most likely this is not a dropped call.

I hope this answers your question

cheers

Edson

View solution in original post

Hello,

Well the CCAPI layer is generally good for protocol's such as SIP and h323 since both these signaling types pass-through this layer in the Symphony system. Depending on the type of connection you have to the PSTN would depend on which debug I would use to monitor certain events. For example if I had PRI/BRI circuit the q931 layer would be enough to look for any suspicious dropped calls on the ISDN trunk. However I would use something else such as the ccsip debugging to troubleshoot sip trunk to the terminating device. Also a simple "show call history voice brief" will give you the disconnect cause code's the previously setup and release calls. The disconnect cause code will give you a hint as to why the call released. For example a cause code (16) would mean that the call cleared normally, as appose to the cause code 38 "network out of order".

I hope this helps

Regards

Edson

View solution in original post

Thanks Edson, I'll keep this in mind.

View solution in original post

8 Replies 8

Edson Pineiro
Level 1
Level 1

Hi Jhun,

To answer your question briefly the following two sections of the 'show call active voice brief' provide information regarding echo and QOS:

noise:-52 acom:45  i/0:-51/-16 dBm

-NoiseLevel=-59

The active noise level for this call.

This value is calculated in the comfort

noise generation module and is used

to generate comfort noise when voice

activity detection (VAD) is enabled.

-ACOMLevel=45

The current ACOM level for this call.

ACOM is the combined loss achieved

by the echo canceler. This value is the

sum of the Echo Return Loss (ERL),

Echo Return Loss Enhancement

(ERLE), and Non-Linear Processing

(NLP) loss for the call.

pl:146400/0ms lost:0/1/0 delay:55/55/85ms

-HiWaterPlayoutDelay=55ms

The First-In, First-Out (FIFO) jitter

buffer high mark that indicates the

maximum depth to which the de-jitter

buffer adapts for this call.

-LoWaterPlayoutDelay=55ms

The FIFO jitter buffer low mark that

indicates the minimum depth to which

the de-jitter buffer adapts for this call.

-ReceiveDelay=85 ms

The current playout FIFO delay plus

the decoder delay for the call.

The following output tells us that there was a codec negotiation using g711ulaw:

g711ulaw  TextRelay: off

Codec negotiation occurs after call setup and alerting states, so most likely the terminating device is off hook.

-be advised that the show call active voice brief only displays active calls, so most likely this is not a dropped call.

I hope this answers your question

cheers

Edson

Hi Edson,

Thank you for the quick response, the information is  a big help. What could be the best command to use to track down drop calls? debug voip ccapi inout mostly i used, aside from this command what else that could give more detailed information of drop calls?

Regards.

Hello,

Well the CCAPI layer is generally good for protocol's such as SIP and h323 since both these signaling types pass-through this layer in the Symphony system. Depending on the type of connection you have to the PSTN would depend on which debug I would use to monitor certain events. For example if I had PRI/BRI circuit the q931 layer would be enough to look for any suspicious dropped calls on the ISDN trunk. However I would use something else such as the ccsip debugging to troubleshoot sip trunk to the terminating device. Also a simple "show call history voice brief" will give you the disconnect cause code's the previously setup and release calls. The disconnect cause code will give you a hint as to why the call released. For example a cause code (16) would mean that the call cleared normally, as appose to the cause code 38 "network out of order".

I hope this helps

Regards

Edson

Thanks Edson, I'll keep this in mind.

Hi Edson,

Is there anyway to differentiate H.323 calls and SIP calls with the fields in the response "show call active voice brief"?

 

Thanks in advance.

Pandi

Hi guys!

  Could someone explain, please, why in some output records the row which starts with Tele 0/2/1:23 (53194) ... is present and in other cases there is no such row (especially when the CallOriginType is Answer ?

If the output has IP then its either sip or h323 call leg. If its tel then
its analog call leg. If neither is present then its call leg invoked with
media resource such as transcoder, conference or mtp. You can confirm this
by issuing the command show sccp conn to see the media resource call legs

Mohammed, thank you for clarification!

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