bearable audible

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Mar 12th, 2007
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One of my customer is complaining that the conversation is clear from US to India, but it is bearable audiable from India to US.


please help.



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mheusinger Mon, 03/12/2007 - 07:26
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Have a look at dropped packets throughout the network. Make sure voip bandwidth is configured properly. Check proper ulaw to alaw conversion.


Without details about the topology it is hard to tell, what is going on. If you use Cisco phones like the 7960, press the "I" or "?" button quickly twice during an affected call at both telephones. It will give you statistics on dropped packets, jitter and delay, which could help.


Regards, Martin

eash Mon, 03/12/2007 - 23:04
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Hi Martin,


thanks for reply,


The setup is follows, There are two 3640 routers connected by 1Mb IPLC link.The entire bandwidth is used only for voice and there is no other traffic.


At india the E1 port of the connects to a dialer and at the remote end the E1 port connects to a PSTN switch at both the ends it is only alaw there is no conversion involved. The calls are made from regular phone are there no IP phones involved.

sometimes the customer complains is echo also in the calls.


the config at both the ends are enclosed.



teodorgeorgiev Tue, 03/13/2007 - 06:42
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Are you using an assymetrical satellite connection?



mheusinger Tue, 03/13/2007 - 08:26
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Hi,


there is no QoS config visible in your routers! This can explain everything. So I would suggest to use "auto qos voip" in your case. Also limit the number of voice cally to the amount of bandwidth you have available. The simplest approach is to change the pri-group to less B-channels.

A configuration could look like this:


hostname POC28


interface Serial0/0

description IPLC Link to India Router

bandwidth 1000 !or whatever bandwidth you really have

ip address x.x.x.x 255.0.0.0

auto qos voip

!


hostname SGT


interface Serial0/0

bandwidth 1000 !or whatever bandwidth you really have

ip address xx.xx.xx.xx 255.0.0.0

auto qos voip

!

Be careful, if the bandwidth is 768k or less, as this would reconfigure the interface to MLPPP resulting in a potential loss of connectivity, if not applied to both ends.


Additionally have a look at the "input gain 2"

and "output attenuation 10" commands. Lowering or increasing attenuation or gain can also result in more volume. This would depend on the signal levels received and I can not tell you which value makes most sense.


One more thing: changing the codec to g729 could also help, as this decreases the amount of bandwidth per call.


Check with "show interface", if there are output drops. This is also an indication of decreased voip quality and should be avoided in any case.


Hope this helps!


Regards, Martin

eash Wed, 03/21/2007 - 05:05
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Hi Martin,


Thanks for the response.


The configs have been changed to include auto qos voip.

The codec used is g729 the capture is enclosed. There are no outout drops.


I have two question


1. how do change the gain and attenuation to correct the low audio.


2. On both the sides only one dial peer for pots and voip is configured. but there are two E1 ports, do we need to change the config.


Regards,

Easwar



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