Conferencing across WAN causes very poor VOIP quality g711u to g729b

[sdoherty]

Hello

When I add a phone across the WAN onto a conference call the quality is terrible.  I did a show sccp conn on the router but do not know what to look for.   Any idea what could be causing the poor quality?  tnks!

67519491   67166757     conf  sendrecv g711u   18146 28076 10.132.222.199
67519491   67166755     conf  sendrecv g711u   16694 28706 10.132.222.192
67519491   67166753     conf  sendrecv g729b   16528 18902 10.132.22.114

COSC-3845VG-161#sh inv
NAME: "3845 chassis", DESCR: "3845 chassis"
PID: CISCO3845         , VID: V03 , SN: FTX1335AJG3

NAME: "c3845 Motherboard with Gigabit Ethernet on Slot 0", DESCR: "c3845 Motherboard with Gigabit Ethernet"
PID: CISCO3845-MB      , VID: V07 , SN: FOC133211JZ

NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 0", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-2MFT-T1/E1  , VID: V01 , SN: FOC13333XTU

NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 1", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-2MFT-T1/E1  , VID: V01 , SN: FOC13333WX7

NAME: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1 on Slot 0 SubSlot 2", DESCR: "VWIC2-2MFT-T1/E1 - 2-Port RJ-48 Multiflex Trunk - T1/E1"
PID: VWIC2-2MFT-T1/E1  , VID: V01 , SN: FOC13333XFA

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 4", DESCR: "PVDMII DSP SIMM with four DSPs"
PID: PVDM2-64          , VID: V01 , SN: FOC13341YY3

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 5", DESCR: "PVDMII DSP SIMM with four DSPs"
PID: PVDM2-64          , VID: V01 , SN: FOC13341YY5

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 6", DESCR: "PVDMII DSP SIMM with four DSPs"
PID: PVDM2-64          , VID: V01 , SN: FOC13341YYH

[Steven Holl]

g729br8 has silence supression built into the codec.  Are you experiencing static and partial audio drops?  Silence suppression can cause that.

You may want to try disable g729br8 on CM and see if behavior changes:

CM Service Parameters> Strip G.729 Annex B> True

Otherwise, you may have packet loss somewhere.  If you invoke a transcoder, there aren't great ways to get RTP stats of the conference bridge.  You'd want to get a packet capture at the conference resource for a bad call and look for packet loss/jitter.  You'd also want to see if any IP phones which hear an issue report packet loss/jitter for the call (check http:// and go to 'Stream 1').