Audio level adjustment on SIP dial peer

[edwardforgacs]

We are trying to reduce the volume on one of the dial peers set up on a CME system.

In IOS version 15.0(1)M+, there are the commands audio outgoing level-adjustment and audio incoming level-adjustment which seem to be what we need. See http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_a1.html.

The problem is, they don't seem to have any effect, even after shutting down the dial peer. There is no perceptible difference in the audio level which is heard through an IP phone even using the maximum settings.

Do these settings only have an effect when the call crosses to the PSTN, or is there something else I'm missing?

[Jonathan Schulenberg]

This was intended for IP-to-IP calls with CUBE. If you have analog or TDM connectivity to the PSTN you should be adjusting the volume on the voice-port with input gain and output attenuation instead.

To answer the actual question for this command: the router needs a DSP to manipulate the audio with the

audio outgoing/incoming level-adjustment command. Assuming you have adequate DSPs installed in the router, you would need to register a transcoding profile to CME.

Please remember to rate helpful responses and identify helpful or correct answers.

[edwardforgacs]

So in the IP to IP scenario, if the call doesn't involve transcoding, the adjustment won't get applied because it never crosses a DSP, correct? Is there some way to force a VOIP dial peer to involve the dial peer?

For example a local IP phone calling the G.711 dial peer, it will never transcode because the phone negotiates G.711.

[Jonathan Schulenberg]

Is there some way to force a VOIP dial peer to involve the dial peer?

You mean force it to use a transcoder? Not to my knowledge. It should invoke it automatically though with those commands present. Be sure that the correct dial-peer is matched "show call actve voice brief".

Please remember to rate helpful responses and identify helpful or correct answers.

[edwardforgacs]

Unfortunately that command alone isn't enough to cause the DSP to be invoked. I have checked it is definately matching the correct dial peer and there is a transcoder set up.

Sample debugs on a call from 101 to 650 using the dial peer 600 shown below:

dial-peer voice 600 voip
translation-profile outgoing ExchangeUM
destination-pattern 6[5-9].
media forking
redirect ip2ip
rtp payload-type comfort-noise 13
session protocol sipv2
session target ipv4:192.168.100.5
session transport tcp
incoming called-number 650
incoming uri from SERVER05
voice-class sip profiles 1
dtmf-relay rtp-nte
playout-delay nominal 100
playout-delay mode fixed
audio incoming level-adjustment -6
audio outgoing level-adjustment -6
codec g711alaw
no vad
supplementary-service h450.12
no supplementary-service sip refer

cme#sh call act voice brief
: ms. () + pid:

  dur hh:mm:ss tx:/ rx:/ dscp: media: audio tos:

media inactive detected: media cntrl rcvd: timestamp:

long duration call detected: long duration call duration : timestamp:
  MODEMPASS buf:/ loss /
   last s dur:/s
FR [int dlci cid] vad: dtmf: seq:
  (payload size)
ATM [int vpi/vci cid] vad: dtmf: seq:
  (payload size)
Tele (callID) [channel_id] tx://ms noise: acom: i/o:/ dBm
  MODEMRELAY info:// xid:/ total://
         speeds(bps): local / remote /
Proxy :

Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
3C36 : 122215 1839501970ms.1 (23:22:27.471 AEST Thu Jan 24 2013) +860 pid:20001 Answer 101 active
dur 00:00:04 tx:211/36292 rx:198/34056 dscp:0 media:0 audio tos:0x0 video tos:0x0
Tele 50/0/1 (122215) [50/0/1.0] tx:4220/4220/0ms g711alaw noise:0 acom:0  i/0:0/0 dBm

3C36 : 122216 1839502480ms.1 (23:22:27.981 AEST Thu Jan 24 2013) +340 pid:600 Originate 650 active
dur 00:00:04 tx:237/37920 rx:225/35872 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 192.168.100.5:23570 SRTP: off rtt:0ms pl:39520/0ms lost:0/0/0 delay:1/0/1ms g711alaw TextRelay: off Transcoded: No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

cme#sh dspfarm dsp active
SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

Total number of DSPFARM DSP channel(s) 0

cme#sh dspfarm dsp all
SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0    1   33.1.0   UP     N/A  FREE  xcode   1      -         -         -
0    7   33.1.0   UP     N/A  FREE  conf    2      -         -         -
0    7   33.1.0   UP     N/A  FREE  xcode   1      -         -         -
0    7   33.1.0   UP     N/A  FREE  xcode   1      -         -         -
0    7   33.1.0   UP     N/A  FREE  xcode   1      -         -         -
0    7   33.1.0   UP     N/A  FREE  xcode   1      -         -         -
0    7   33.1.0   UP     N/A  FREE  xcode   1      -         -         -

Total number of DSPFARM DSP channel(s) 7

[Aaron Harrison]

What's the basic problem? I.e. why do you need to reduce the volume?

[edwardforgacs]

The Voicemail system generates audio which is disproportionately loud to everything else on the network or the PSTN and there is no way to reduce the volume on the voicemail system itself.

If the audio level adjustment commands could be made to work in the way it looks like they are intended, it would fix the problem perfectly.

[Aaron Harrison]

That's interesting. Have you manipulated the audio levels of other stuff on the network to get into this situation?

I have several customers using Exchange UM and have never had this issue.

[edwardforgacs]

Not at all. I guess it's a bit of a minor point, but the volume of Exchange UM (2007 & 2010) has always been too loud.

[Jonathan Schulenberg]

Ok, realizing this doesn't answer the still-valid original question, here's a thought to consider: Is Exchange UM too loud or is the PSTN too quiet and everyone has the volume of their phones cranked way up? I see this *a lot* where the volume is set to 90% or 100% on the phones so that the PSTN sounds normal and then an IP application, Exchange or otherwise, sounds like a bullhorn. The phones have an amplifier on the analog side of the DSP so setting volume to 100% is orders of magnitude higher than the original input volume.

[edwardforgacs]

Thanks, yes, I understand your point but the most phones' volume are not above 50%. If anything the audio from our ISDN is also quite loud, but not bordering on uncomfortable like Exchange. The ISDN is actually "too loud" according to some fax tests I have been doing.

The point is that if we reduce the ISDN gain or increase attenuation on the PRI port then the difference between Exchange and it will be a lot more pronounced.