Increase E&M voice transmit delay

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Aug 19th, 2010
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This may sound weird but does anyone know how I can increase voice transmission delay on Cisco 3600 series router (IOS c3640-is-mz.123-25.bin) analog E&M port?


I have a working connection trunk with the config below. The connection trunk interfaces to old paging equipments.


Thanks.

Router A

voice-port 3/1/1
operation 4-wire
type 5
signal immediate
no echo-cancel enable
no comfort-noise
cptone MY
timeouts call-disconnect 3
timeouts wait-release 3
connection trunk 4042311
!

dial-peer voice 4042 voip
destination-pattern 4042...
session target ipv4:10.230.42.2   <--- Router B
dtmf-relay h245-alphanumeric
codec g711alaw
ip qos dscp cs5 media
ip qos dscp cs5 signaling
no vad
!
dial-peer voice 4240111 pots
destination-pattern 4240111
port 3/1/1
!

Router B

voice-port 1/1/1
operation 4-wire
type 5
signal immediate
no echo-cancel enable
no comfort-noise
timeouts call-disconnect 3
timeouts wait-release 3
connection trunk 4240111
!

dial-peer voice 4240 voip
destination-pattern 4240...
session target ipv4:10.230.40.18    <--- Router A
dtmf-relay h245-alphanumeric
codec g711alaw
ip qos dscp cs5 media
ip qos dscp cs5 signaling
no vad
!
dial-peer voice 4042311 pots
destination-pattern 4042311
port 1/1/1

!

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Overall Rating: 3 (1 ratings)
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paolo bevilacqua Thu, 08/19/2010 - 01:18
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Transmission delay ?

On a voice port ?

What are you talking about ?

pllim Thu, 08/19/2010 - 01:51
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I would like to know if its possible to delay receiving of voice traffic at router B when it is sent from router A? In otherwords, is it possible to increase the latency delay?

paolo bevilacqua Thu, 08/19/2010 - 02:04
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No, it is not possible to do that.


You can change the codec packet size but that will only affect the delay in a non perceivable manner.

Aaron Harrison Thu, 08/19/2010 - 02:18
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Hi


Perhaps try to describe the problem you are trying to solve?


Tranmission delay in VoIP solutions is something that is always being minimised as it is only ever the source of problems; not something you would ever want to delay intentionally.


Regards


Aaron

paolo bevilacqua Thu, 08/19/2010 - 02:30
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I also wanted to ask the same but refrained in this case.


In telephony, often clients ask for the strangest things that only makes sense to them.


Being this a PA connection, I can try a long shot: After speaking into phone, the client wants the time to walk to a window and hear his announcement from the PA speakers ?


Whatever the reason, if interested in a custom script that inserts a programmable delay in PA messages, or any other desired functionality, you can contact me at the address present in my profile.

pllim Thu, 08/19/2010 - 02:47
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Ok, let me tell you the reason for doing this.


I need the delay because I am trying to synchronize the link from Site A (Router A) -> Site B (Router B) with other links (from Site A to Site C, D, ...) that are not using E1 line. The other links/sites are using the low speed 4-wire analog leased line without router. E1 line is already available at Site A+B location and I am trying to see if I can make use of the E1 instead of the existing analog leased line.


All the Paging equipments at the sites need to synchronize together. To illustrate in a diagram -


Master Paging Equip. -- Router A ------------------(E1)--------------------- Router B -- Paging Equip. B

            |    |

            |    |

            |    |------------------------------(analog leased line)------------------------------------Paging Equip.C

            |

            |-----------------------------------(analog leased line)------------------------------------Paging Equip.D

pllim Thu, 08/19/2010 - 04:02
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Btw, the Paging Equipment at sites are actually Paging Transmitters. The Transmitters purpose is to broadcast text messages to pagers. The location of the sites are between 5km - 50km from Site A. Just thought you should know this.

Aaron Harrison Thu, 08/19/2010 - 04:08
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OK... I think you will struggle a lot with this.


Is this all theoretical or have you tried this and verified it works faster over an E1?


Generally it works the other way round; delays are bigger when running over VoIP due to packetisation/network delays and generally this causes  things to work slower and less reliably at high speeds. For example, you take a fax off a POTS line and put it on a voice gateway... you lose high/Super G3 speeds, get problems with error correction, and on a well configured system you always expect faxes to be notably slower than the equivalent POTS line.


I'm assuming here that the connection is working as an modem-type connection?


Aaron

pllim Thu, 08/19/2010 - 19:31
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Well, I am not able to verified whether E1 is slower or faster. The only thing that I could see from an analyzer is that the E1 transmission is not synchronize with the rest of the analog line transmissions.


I'm just suspecting that the E1 is faster since it is using fiber optics cable and bandwidth is higher compare to low bandwidth analog 4-wire copper cable.


Yes, the analog line transmissions are working like a modem-type connection.


If what you are saying is true then I should be looking at the other direction - how to reduce the latency delay  (or any other delays). I have pasted result of a show call active voice command. Is there anything that I can optimize?

 

#show call active voice


GENERIC:
SetupTime=1295578114 ms
Index=1
PeerAddress=4240111
PeerSubAddress=
PeerId=4240
PeerIfIndex=22
LogicalIfIndex=0
ConnectTime=1295578144 ms
CallDuration=00:27:09 sec
CallState=4
CallOrigin=2
ChargedUnits=0
InfoType=speech
TransmitPackets=81848
TransmitBytes=13045280
ReceivePackets=81810
ReceiveBytes=13039200
VOIP:
ConnectionId[0xF45ACB79 0xAB3411DF 0x99C9EA04 0x7D4B1FAA]
IncomingConnectionId[0xF45ACB79 0xAB3411DF 0x99C9EA04 0x7D4B1FAA]
RemoteIPAddress=10.230.40.18
RemoteUDPPort=16782
RemoteSignallingIPAddress=10.230.40.18
RemoteSignallingPort=13691
RemoteMediaIPAddress=10.230.40.18
RemoteMediaPort=16782
RoundTripDelay=37 ms
SelectedQoS=best-effort
tx_DtmfRelay=h245-alphanumeric
FastConnect=TRUE

AnnexE=FALSE

Separate H245 Connection=FALSE

H245 Tunneling=TRUE

SessionProtocol=cisco
ProtocolCallId=
SessionTarget=
OnTimeRvPlayout=1626680
GapFillWithSilence=0 ms
GapFillWithPrediction=720 ms
GapFillWithInterpolation=0 ms
GapFillWithRedundancy=0 ms
HiWaterPlayoutDelay=65 ms
LoWaterPlayoutDelay=55 ms
TxPakNumber=81376
TxSignalPak=0
TxComfortNoisePak=0
TxDuration=1627520
TxVoiceDuration=1627520
RxPakNumber=81338
RxSignalPak=0
RxDuration=0
TxVoiceDuration=1627450
VoiceRxDuration=1626680
RxOutOfSeq=36
RxLatePak=0
RxEarlyPak=2
PlayDelayCurrent=55
PlayDelayMin=55
PlayDelayMax=65
PlayDelayClockOffset=1847192895
PlayDelayJitter=0
PlayErrPredictive=720
PlayErrInterpolative=0
PlayErrSilence=0
PlayErrBufferOverFlow=30
PlayErrRetroactive=0
PlayErrTalkspurt=0
OutSignalLevel=-67
InSignalLevel=-60
LevelTxPowerMean=0
LevelRxPowerMean=-607
LevelBgNoise=0
ERLLevel=20
ACOMLevel=20
ErrRxDrop=0
ErrTxDrop=0
ErrTxControl=0
ErrRxControl=0
ReceiveDelay=55 ms
LostPackets=36
EarlyPackets=2
LatePackets=0
VAD = disabled
CoderTypeRate=g711alaw
CodecBytes=160
Media Setting=flow-through
CallerName=
CallerIDBlocked=False
OriginalCallingNumber=4240111
OriginalCallingOctet=0x80
OriginalCalledNumber=4042311
OriginalCalledOctet=0x80
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=4240111
TranslatedCallingOctet=0x80
TranslatedCalledNumber=4042311
TranslatedCalledOctet=0x80
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=4042311
GwReceivedCalledOctet3=0x80
GwReceivedCallingNumber=4240111
GwReceivedCallingOctet3=0x80
GwReceivedCallingOctet3a=0x0
Username=

GENERIC:
SetupTime=1295578124 ms
Index=1
PeerAddress=4042311
PeerSubAddress=
PeerId=4042311
PeerIfIndex=23
LogicalIfIndex=16
ConnectTime=1295578124 ms
CallDuration=00:28:04 sec
CallState=4
CallOrigin=1
ChargedUnits=0
InfoType=speech
TransmitPackets=84528
TransmitBytes=13472496
ReceivePackets=84576
ReceiveBytes=13480176
TELE:
ConnectionId=[0xF45ACB79 0xAB3411DF 0x99C9EA04 0x7D4B1FAA]
IncomingConnectionId=[0xF45ACB79 0xAB3411DF 0x99C9EA04 0x7D4B1FAA]
TxDuration=1684300 ms
VoiceTxDuration=1684300 ms
FaxTxDuration=0 ms
CoderTypeRate=g711alaw
NoiseLevel=-59
ACOMLevel=20
OutSignalLevel=-67
InSignalLevel=-58
InfoActivity=2
ERLLevel=20
SessionTarget=
ImgPages=0
CallerName=
CallerIDBlocked=False
AlertTimepoint=1295578124 ms
OriginalCallingNumber=4240111
OriginalCallingOctet=0x80
OriginalCalledNumber=4042311
OriginalCalledOctet=0x80
OriginalRedirectCalledNumber=
OriginalRedirectCalledOctet=0xFF
TranslatedCallingNumber=4240111
TranslatedCallingOctet=0x80
TranslatedCalledNumber=4042311
TranslatedCalledOctet=0x80
TranslatedRedirectCalledNumber=
TranslatedRedirectCalledOctet=0xFF
GwReceivedCalledNumber=4042311
GwReceivedCalledOctet3=0x80
GwReceivedCallingNumber=4240111
GwReceivedCallingOctet3=0x80
GwReceivedCallingOctet3a=0x0
GwOutpulsedCallingNumber=4240111
GwOutpulsedCallingOctet3=0x80
GwOutpulsedCallingOctet3a=0x0

paolo bevilacqua Fri, 08/20/2010 - 00:34
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In my opinion, The E1 VoIp circuit will always be slower:


- All circuits are using modems, supposedly low speed ones. Transmission delay is unknown, let say value D.

- On top of D, the VoIP circuit on E1 adds codec / packetization delay (some 20 ms), and transmission delay (2-4 msec the most).


Consequently, the VoIP E1 circuit is slower.

You can change the sample size to 80 bytes, that lowers packetization delay to 10 msec, that is still "not synchronized" probably.


This said however, my understanding of synchronization in multiple repeater radio systems, is that synchornization is necessary only in areas where there can be an overlap of signals into a same receiver. Consequently, if an area is served by single repeater, there is no possibility on interference and some time shift from source signal does not cause problems. May be that is the case with the area served by the E1 circuit, you can check that.

pllim Fri, 08/20/2010 - 01:51
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Yes you are right. The synchronization is needed because we have multiple areas, each covered by a paging repeater. When synchronized, the overlapping areas are known to us.


Besides changing the sample size to 80 bytes, are there any other ways to reduce transmission delay? What about reducing playout-delay?

paolo bevilacqua Fri, 08/20/2010 - 02:29
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There is no other way or I would have mentioned it.


I guess you will have to re-design this from ground.


Please remember to rate useful posts clicking on the stars below.

paolo bevilacqua Fri, 08/20/2010 - 03:01
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Thank you for the stellar rating of "3".


It is quite disappointing to see how often people that doesn't get the answer they want, will judge posts as not being useful, no matter if they are 100% correct and factual, and made with the best will to help.


Fortunately, not everyone behaves so.

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