Active call suffers when another call is coming in/active

Nishant Jethwa · ‎09-16-2010

When there is an active call, if another call comes in, the exisiting call suffers, crackly and broken voice.

Site configured with it's CUCM location set to to 120kbps for audio. From what I understand, this controls call in both directions.

I believe incoming calls come come over their WAN link from their main site.

In terms of how busy this site is, I don't believe they have more than 3 calls on the go.

Questions:

How do I see how many inbound calls are up?

What could be causing existing calls to suffer when new calls come in?

Aaron Harrison · ‎09-16-2010

Hi

Basically a 120Kbps location means that if you are running G729, CCM will permit 5 calls in or out of that location concurrently (CCM works it out as 5*24 = 120Kbps). If you run G711, then the system will permit 1 call (80*2=160, so the second would be blocked).

If you have OK quality, then another call starts and the quality drops, this generally means that:

- The line is congested, but your QoS on the WAN is protecting your voice traffic OK until..

- The new call starts, and current amount of EF bandwidth going over the WAN now exceeds what is provisioned

Depending on your setup, it may be that it is the remote site's EF allocation that is exceeded, or it could be the far end (e.g. the head office).

I would first confirm how much bandwidth you have on the WAN, and verify this matches your locations configuration - bear in mind the calculations above, which is what CCM uses as it doesn't really know what bandwidth your calls may use.

For example - on a serial link the bandwidth used for a G729 call may be much lower than the bandwidth used on an Ethernet presentation. Work out how many calls you should get over the WAN, and multiply it by 24 if G729, or 80 if G711 to work out what to put in locations.

Alternatively you can check in RTMT the performance counters - there are counters for 'available bandwidth' and so on for the locations. You can look at how many are running at once, and then have the staff set up new calls until problems occur. When they do, reduce the locations bandwidth by one call, and then try again.

What will happen is that CCM will block the calls when the limit is reached; this should prevent quality issues, but if they need to be making more calls you will still have a problem. You may need to increase the available EF bandwidth on the WAN, or introduce AAR to dial around the WAN.

Regards

Aaron

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Nishant Jethwa · ‎09-16-2010

Hi Aaron,

Thanks for the indepth reply. Definitely clears up any confusion on how the system works.

There was a QOS policy (service-policy output) applied on the Multilink which was then applied to the ATM interface via a Virtual Template.

It was 60, changes this to 120

However, there doesn't seem to be much change.

Had a look at the locations stats (CallsInProgress, BandwidthAvailable, BandwithMaximum)

With 2 calls, bandwidth seems to be maxing out at 120 and the Bandwidth Maximum is constanlty on 120. The region settings are all set to g729

Current QOS policy is below

access-list 10 permit 10.x.x.128 0.0.0.127

class-map match-all Voice

match access-group 10

policy-map Dscp

class voice

priority 120

class default

Aaron Harrison · ‎09-17-2010

Hi

What type of circuit is this? Is it a link into a service provider MPLS network, or a private/point-to-point link?

If it's private, you must increase the bandwdith in both directions (to and from the site, so at the remote end of the link in the transmit direction).

If it's MPLS, you have to get the SP involved otherwise you still risk congestion.

Regards

Aaron

Aaron Please remember to rate helpful posts to identify useful responses, and mark 'Answered' if appropriate!

Nishant Jethwa · ‎09-20-2010

It's an ADSL circuit point-to-point setup.

I had a look at the bandwidth setting for intersite calls, they're all set to g.729 how they should be.

Odd thing is, I did a rest on the router and according to RTMT, it shoots up to 2 calls up as soon as the router comes up (I checkd other site and they all seems to be running on a basline of 0 unlike this site which seems to be 2).

I'm going to create a new DP, region and location and see if there is something wrong with the callmanager setup.

From debugging the router, is there a command that can show the voip path of the incoming call?