3 way calling sound like underwater

[sdwitner1]

I have an office with a 2811 voice gateway and 2960 switch and 5 7942 IP Phones. When the manager is on a 3 way conference call which he originates from his IP phone the callers on the remote end say he sounds like he is under water. They have a full T1 with QoS. The T1 never gets over 60% utilization and there are no errors on the T1.

[Jonathan Schulenberg]

Does the problem persist if all three call participants are local in the same office?  If so, does it persist if you use a different conference bridge resource (e.g. CUCM software bridge vs. IOS HW bridge)?

These steps would rule in/out WAN QoS and a buggy or defective software load on the DSP or IPVMS respectively.

[sdwitner1]

here is when they notice the issue. When the manager initiates a conference between his IP phone in the office and 2 outside clients it's fine. The issue surfaces when anybody else in the office makes a call from their IP phones while the managers conference call is in place. Then the outside clients will notice that the manager sounds like he is talking under water.

[sdwitner1]

here is the conferencing config in the 2811 router....

voice-card 0

dspfarm

dsp services dspfarm

!

sccp ccm group 1

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 1 register McCabe_CFB

registration retries 32

connect retries 10

connect interval 30

!

dspfarm profile 1 conference

codec g711ulaw

maximum sessions 4

associate application SCCP

[Jonathan Schulenberg]

The issue surfaces when anybody else in the office makes a call from their IP phones while the managers conference call is in place. Then the outside clients will notice that the manager sounds like he is talking under water.

Well now this is a QoS problem. Look at your policy-map; packets are likely being dropped from the LLQ. Remember that the LLQ is policed at the rate specified. It appears you only have enough bandwidth to support three calls; or, the conference call (which is sourced from a different IP than the phone) is not getting placed into the LLQ class.

Be careful to know which codec the calls are at when doing your math. For example, if you have four G.711/722 with 20msec sampling would require at least 352kbps at layer two (assuming MLPPP and RTCP). You can use http://www.bandcalc.com/ to take some of the guess work out of this.

You may want to include the sccp local Loopback0 command to ensure you know which interface confernce calls will be sourced from.

[sdwitner1]

class-map match-any VOICE-CALL-SIGNALLING

match dscp cs3

class-map match-any DEFAULT

match dscp default

class-map match-any VOICE-BEARER

match dscp ef

!

policy-map WAN-EDGE

class VOICE-BEARER

  priority percent 33

class VOICE-CALL-SIGNALING

  bandwidth percent 5

class class-default

  fair-queue

!

interface serial0/0/0

ip address x.x.x.x x.x.x.x

encapsulation ppp

max-reserved-bandwidth 100

service-policy output WAN-EDGE

[sdwitner1]

all my voice is delivered via SIP trunks to the router. The ISP has the SIP trunking configured for 5 concurrent calls.