Re: Voice Quality issue

s.nicholls · ‎02-07-2003

We have a two pabx's connected together via Cisco 3600 Routers using VOIP and Dial peers The link is 2mg .A user at the remote site will make a call which breaks out into the PSTN via the local site,but every now and then this call will be of such a poor quality that it needs to be terminated.

Qos is set up and there appears to be no bandwidth issues or packet loss.

It looks like it only happens on digital phones though

Does anyone know how to fault find this?

csmith · ‎02-11-2003

Is the 2mgs for voice only, if it's sharing the pipe with data, are you using LFI?

s.nicholls · ‎02-11-2003

The link carries data as well and Qos is set up to give 700k Guaranteed but no LFI is not used.

Would it help?

csmith · ‎02-11-2003

Without looking at a sniffer I can't be sure, but it is a possability. LFI fragments your data so that large ethernet packets don't take up to much seralization time on the interface. It is possable to have LLQ and still have problems with large packets.

It's just a thought.

check this out:

http://www.cisco.com/warp/public/732/Tech/link/

s.nicholls · ‎02-11-2003

Thanks .Does this mean that even though we prioritise VOIP traffic does this mean that a large packet can affect the voice.It only appears to be a one way problem .Speech from the remote user (PSTN SIDE) is dropped ie if you count 1,2,3,4,5,6 you may not hear 3,4

csmith · ‎02-11-2003

I think I need more info...

Setup:

Phone - PBX - Router - WAN - Router - PBX - Phone

If the troubled call is from the PSTN only then you need to determine the path of the voice coming from the PSTN. (What equipment is involved) If you have a PRI or Trunk locally then it's a PSTN problem not a VoIP. Does the PBX on both sides have it's own PSTN access? If so it's on a VoIP problem it's a PBX or telco issue.

Hope this helps.

s.nicholls · ‎02-11-2003

it only happens on PSTN calls which are all routed across the 2mg link.Internal calls across the link are not affected (as far as we can tell)

Jay-Tucson · ‎02-13-2003

What is your voice compression method and bandwidth?

s.nicholls · ‎02-14-2003

Compression is G729r8 Bandwidth is 2.048 mg with 700k reserved for the voice.

michelbowman · ‎03-16-2003

I am experiencing an identical issue .. but my routers are connected directly via E1- PRI's to the PSTN on one end and via T1's on the other end ..I too have an E1 in this case 1Meg is dedicated to VoIP .the baffling issue is that one side can hear

the other end clearly 100% of the time and the other end has the worst quality 70-90% of the time. Is there a setting that can be tweeked to make this issue go away ? It seems strange that only one side experiences the quality issue.

Any ideas or reasons why this is happening ?

thanks in advance

Mich

s.nicholls · ‎03-17-2003

It looks like it could be to do with VAD (Voice Activation Detection).This can cause problems across the PSTN.

We have tried turning this off and found that the voice quality improves,however the CPU Utilisation on the 3620 goes to 100%.

So we then changed the sampling rate on the Codec from 20k to 60k and this cured the CPU problem.

This has all been done in a test enviroment ,but I mean to implement it on the live system this week

Steve

fjahangir · ‎03-24-2003

I am also using 2610 router both end and voip. QOS is enabled, one end voice=48 kb other end 32 kb for voice. We are using remedy application and connected to US pbx simutanously but facing one end reciving time low voice quality. If this solution which you are going to implement works inform me. What do you mean by Codec and how to increase tell me.