'voice-class codec' for SCCP phones (CME)?

Hi, with SIP phones it's possible to apply a codec voice class.
Let's say I have the following voice class:
voice class codec 1
codec preference 1 g722-64
codec preference 2 g711alaw
I can apply it for SIP phones, e.g. for pool 9:
voice register pool 9
voice-class codec 1
With SCCP phones, I can only set one codec with the 'codec' command under ephone.
My goal is to use 'codec transparent' in the dial peer and to let the phone itself negotiate the codec. How can I do this with SCCP phones?
For example, if I use 'codec transparent' in the dial-peer and someone (who doesn't support g722) calls me, then the SIP phone negotiates g711alaw with the other side and no transcoding is needed. This is what I also want for my SCCP phones. Am I missing a command?
I'm using CME 8.6

The syntax for SCCP phones is the same. Just apply the class to the VoIP dial peers.
dial-peer voice 100 voip
tone ringback alert-no-PI
description For InBound VoIP
modem passthrough nse codec g711ulaw
voice-class codec 1 <<<<<
voice-class h323 1
incoming called-number .
fax rate disable
no vad
Please rate helpful answers!

Similar Messages

Voice class codec issue

Hi,
I have a sip trunk terminating on a CUBE. On the CUBE, I hard-code the dial-peer 211 with G.729 codec. On an inbound call to a DID terminating on the IP phone, the call negotiated g.729 and completed successfully. No issues.
Now I created a voice class codec with 2 codecs in the following preferences
voice class codec 1
codec preference 1 ---> g729
codec preference 2 ---> g711
I then apply this to a dial-peer 211.
On an inbound call from the same incoming number to the same DID(same endpoint), this call is now negotiated only at G.711 codec. I verified that I am hitting the same dial-peer by using "show call active voice brief" and checking the pid. It is using dial-peer 211.
My expectation is that the call will still negotiate G.729 and will use G.711 only if the call cannot be completed at G.729.
As a test, I also verified by removing codec preference 2 (i.e.) G.711 from voice class codec 1 and call negotiated at G.729.
BTW, in each scenario, I used the show voice call active compact to verify the call legs and codecs being used.
CUCM version 9.1 and IOS 15.1(4). Any ideas why this odd behavior?
Regards,
K iyer

Here is the catch with CUCM. CUCM always prefers and will use the "best available codec" offered. Therefore, when the gateway forwards the setup to CUCM, it would see g711 as a valid option and would use it. Here is more info on the same:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/9_1_1/ccmsys/CUCM_BK_C5565591_00_cucm-system-guide-91_chapter_0101.html#CUCM_RF_RE6237E1_00
Regions
Regions provide capacity controls for Cisco Unified Communications Manager multi-site deployments where you may need to limit the bandwidth for individual calls that are sent across a WAN link, but where you want to use a higher bandwidth for internal calls. Additionally, the system uses regions also for applications that only support a specific codec; for example, an application that only uses G.711. Use regions to specify the maximum transport-independent bit rate that is used for audio and video calls within a region and between regions; in this case, codecs with higher bit rates do not get used for the call.
Cisco Unified Communications Manager prefers codecs with better audio quality. For example, despite having a maximum bit rate of 32 kb/s, G.722.1 sounds better than some codecs with higher bit rates, such as G.711, which has a bit rate of 64 kb/s.
HTHs
Please rate helpful posts.

Voice-Class Codec question

What is it meant by allowing g711 30ms and g729 60ms.. I know this has to do with the bytes per frames, but how would you put that in the voice-class codec command.. How do you come up with the the total of bytes to accomplish the task...?

i didt get your question,do you mean this?
voice class codec 101
codec preference 1 g729r8 bytes 40
codec preference 2 g723r63 bytes 48
codec preference 3 g723ar63 bytes 48

Polycom sip phone cme

has anyone been able to register a polycom sip phone on a callmanager express? i especially need help on configuring the polycom phone

I just got this to work with a SoundPoint 501. Took some fiddling, but phone is registered and seems to work Ok. Still working on some issues with WMI and getting the messages button to work correctly, but the phone is registered.
here is the config:
voice register global
mode cme
source-address 10.72.13.19 port 5060
max-dn 200
max-pool 20
timezone 13
tftp-path slot0:
create profile sync 0390651099874124
ntp-server 10.71.13.19 mode directedbroadcast
voice register dn 1
number 4020
allow watch
voice register template 1
session-transport udp
softkeys hold Resume Newcall
softkeys idle Newcall Redial Cfwdall
softkeys connected Endcall Trnsfer Confrn Hold
voicemail 4200 timeout 30
voice register dialplan 1
type 7940-7960-others
pattern 1 4...
pattern 2 ....
pattern 3 .
voice register pool 1
id mac 0004.F213.2465
type P600
number 1 dn 1
dialplan 1
dtmf-relay rtp-nte
voice-class codec 1

HT1349 What is the best app for voice caller ID for Iphone 4s

Waht is the best app for voice caller ID for I Phone 4s

Please post to the iPhone forum, not the Enterprise forum.

How to config CME with FXS card for analog phones

I want to config FXS cards in CME router with SCCP protocol for transfer, pickup ... feature. I can find information about the sccp (the command about stcapp), but I can not find the information about how to config CME, e.g. the phone no.(something like ephone-dn .... )
Can anyone help me ? or send me the link
Thx
Dennis

Thx Johncaston,
Your config is for H323, it doesn't support features, something like transfer funtion .......
I have found the config for SCCP, let's share my finding
====================
ephone-dn 178
number 9933
description Board Room
name Board Room
hold-alert 30 originator
ephone-dn 179 dual-line
number 9990
description Meeting Room 3
name Meeting Room 3
hold-alert 30 originator
ephone 95
mac-address 9556.60FA.0200
type anl
button 1:178
ephone 96
mac-address 9556.60FA.0201
type anl
button 1:179
GigabitEthernet0/0 is up, line protocol is up
Hardware is MV96340 Ethernet, address is 0019.5566.0fa0 (bia 0019.5566.0fa0)
================================
stcapp ccm-group 1
stcapp
sccp local GigabitEthernet0/0
sccp ccm 172.16.136.125 identifier 1
sccp
sccp ccm group 1
associate ccm 1 priority 1
signaling dscp af31
audio dscp ef
voice-port 1/0/0
cptone HK
caller-id enable
voice-port 1/0/1
cptone HK
caller-id enable
dial-peer voice 21 pots
service stcapp
port 1/0/0
dial-peer voice 22 pots
service stcapp
port 1/0/1
=========================
Thanks !!
Dennis

Building CME with SIP vs. SCCP Phones

We currently have 4 CME Systems with SCCP phones. Recently, it was discuss by our upper management to start using SIP phones. Right now the system I have configured are using a PRI for dialtone with POTS lines for backup. What needs to different on the config to use SIP? Do I replace the 'telephony-service' and current 'dial-peers' or add to them?
This is all new to me and I still have not been able to attend any formal training.
Would there ever be a reason to use SIP and SCCP phones on the same system and would that even work?
Any help is appreciated!

Aha.... that's a different question.
The choice of your trunking protocol (PRI versus SIP versus something else like FXO/analogue) is largely independent of the protocol the handsets use. You can use SCCP for your handsets and then use SIP trunking into a provider for your external calls. The CME Router will handle the signalling translation between SCCP and SIP for you.
Like everyone else here, I would always use SCCP for the handsets. Whilst Cisco are improving SIP handset features all the time between releases you still can't do everything you that you can do with SCCP. You would also need version 15 to support some of the more useful SIP handset stuff including Extension Mobility, and my own personal view of the code quality on the version 15 trains would use such language that it can't be mentioned here.
SCCP to the handset and trunking over SIP is perfectly valid and I've used it many times with great success.
Hope this helps. Barry

Support problem. The adviser tried to diagnose the failure of an application called JOTNOT PRO to update. He said he'd phone back at 8pm. I pick up his call at 8pm. Electronic voice: thank you for calling apple. We are now closed" So now I have 2 problems

Support problem. The adviser tried to diagnose the failure of an application called JOTNOT PRO to update. He said he'd phone back at 8pm. I pick up his call at 8pm. Electronic voice: thank you for calling apple. We are now closed" So now I have 2 problems: the adviser had got me to reset so bye bye wifi passwords and settings. And then when he phones back as we agreed he's not actually on the phone, it's just a voice saying Apple Support is now closed! What on earth is going on at apple support I wonder.
Anyway, the original problem: an application called JotNot Pro fails to update. The error message says something like "you cannot update this cos either you bought it with a different Apple ID or somebody else bought it."
Neither applies. Can anybody help?
PS. This is my first approach to the support community so please bear with me!
William

You might have better luck contacting the app's support or developers.

Create cnf-files (sccp) for 7912G phone appears to be wrong

Hi again Cisco techs,
Have added file for 7912G phone.
Uploaded 7912 sccp firmware for 7912G phone via tftp-server and sh running-config confirms this
However. run create cnf-files shows differently. Same as sh telephony-service tftp-bindings shows 7960 files
Confirmed files via create cnf-files command
System replies with below error
CNF file creation is already On
Updating CNF files
%Error deleting flash:SEPDEFAULT.cnf (No such file or directory)
%Error deleting flash:XMLDefault.cnf.xml (No such file or directory)CNF-FILES: Clock is not set or synchronized,
retaining old versionStamps
CNF files update complete
cme_router(config-telephony)#
cme_router#
Have attached output of more sepdefault.cnf and xmldefault.cnf.xml and sh telephony-service tftp-bindings for analysis.
I suspect that first file is corrupt and 2nd file needs updating with correct information
Can you confirm this please?
Please advise if you can.

That did not work, but using CS6 instead of CS5 seemed to do the trick. Perhaps I tried to use the old version on newer projects, and the program simply pronounce the file as corrupt.
Thanks for the comebacks.

What is the best Voice recognison for calling phone application

What is the best Voice recognison for calling phone application in the market now, free or not ????
Do you have any sugesstion

In my Opinion.. Vlingo (Free) is the best voice rec phone dialer... It is the swiss army knofe of voice recognition for the iPhone.

How do i monitor SIP Phone from a SCCP Phone? UC500

Hello Community.
This is my SIP Phone config:
voice register dn 2
number 214
name xxx
no-reg
label xxx
voice register pool 2
id mac 44E4.D944.9B11
session-transport tcp
type 9971
number 1 dn 2
number 2 dn 3
dtmf-relay rtp-nte
username xxx password xxx
codec g711ulaw
And this is the SCCP Phone:
ephone 12
device-security-mode none
mac-address 0011.2014.8248
ephone-template 16
username xxx password xxx
type 7960 addon 1 7914
button 1:12 2:51 8m11 9m10
button 10m15 11m13 12m16 13m17
button 14m18
Hope you can help, thanks Patrick

Yes you can.
Sample config :
sip-ua
          presence enable
     presence
          presence call-list
          max-subscription 64
          watcher all
          allow subscribe
     ephone-dn 12
          number 2010
          allow watch
          blf-speed-dial 1 214 label "Teset-User"
     voice register dn 2
          number 214
          allow watch
CME admin guide:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmepres.html#wp1011193
Please rate replies and mark question as "answered" if applicable.

Total Bandwidth req for IP Phone

Hi,
consider the scenario:
I have centralized deployment.I have branch site which will make a call in ratio of 2:1 (if there is 2 phones one will be active ONcall).
I want to know
1.what will be the Bandwidth required for SCCP registration on CCM for single IP Phone
2.What will the bandwidth requirement for Keepalive messages between branch and data centre for single IP Phone with MGCP as wellas H323
3.Will there be any other thing utilizes the bandwidth if so what are they?
Thanks,
Hari

the main thing u need to consider in voip bandwidth
is the codec in use
for example with g711 voice call consude about 64k while with g729 24 this is whiout over head
one more idea
if u are using CAC with CCM one call g711 needs 80k and g729 24k
while with an IOS gatekeeper
g711 needs 128k and g729 16k per call
all above without L2 overhead
good luck
if helpful Rate

What licenses do I need for CUCM and CME

Hi Experts,
I am a newbie to Cisco IP telephony and request your guidance.
My boss has asked me to order some 7942 phones for a CUCM based site and some 7942 phones for a CME based site.
For CUCM 9, I understand that we need an enhanced user licenses on CUCM side.
For the IP phone, I see 2 part numbers and am confused.
CP-7942G - Cisco Unified IP Phone 7942G
CP-7942G-CH1 - Cisco Unified IP Phone 7942G, for Channels, with one station user license
What is the station user license mentioned in the part number CP-7942G-CH1 ? Is it the same as the enhanced user licenses? Assuming i am ordering enhanced user licenses, what should I order CP-7942G or CP-7942G-CH1?
I also see a part number SW-CCM-UL-7942= ? What is this part? is it needed?
And what do I need for CME?
Is FL-CME-SRST-10 enough for the CME?
What is the part number for the phones? Is it CP-7942G or CP-7942G-CCME?
I also see another part number "SW-CCME-UL-7942="? What is it? And is it really needed?
I heard that Cisco simplified Licensing from version 9. At least it doesn't look simple to me :(
Can you please help me, Experts?
Thanks,
Pete

Hi without knowing your business goals, current network, etc here is some feedback.
Depending on your design, if the sites are connected via a reliable WAN you could run the branch site as SRST and not CME, this way you get a rich feature set of the centralised call manager and simplify licenses, etc.
Anyway, you need to size your CUCME (or SRST) correctly as each Cisco router hardware platform has a maximum supported handsets. If you have a SIP provider new purchased ISRs now come with 10 CUBE licenses other wise you will need something else such as a VIC for dial tone.
For instance the following is a router that comes with 25 CME
C2901-CME-SRST/K9
2901 Voice Bundle w/PVDM3-16,FL-CME-SRST-25,UC Lic,FL-CUBE10
You can add CP-7942G-CCME which is the physical phone and license for CME
Use CP-7942G= for the Call Manager Deployment as you will get UCL Enh or CUWL for the user side
If you have existing routers you will need to obtain upgrade licenses for voice/CME add DSPs if needed for conferencing, media termination, etc, CUBE or voice cards and you can add the phones.
Router handset capacities see Table 7
http://www.cisco.com/c/en/us/products/collateral/routers/2900-series-integrated-services-routers-isr/data_sheet_c78_553896.html
Hope this helps

Sccp phones unregistred

Hi all
I am upload 7937 firmware in telephony-service and recreate cnf files .
After that when restart the SCCP phones they can't registred in CME but receive DHCP IP addresses and CME IP .
I execute no load 7937 firmware and perform no cre cnf , cre cnf in telephony-service but its not helps me. Phones who not restarted working fine
sh telephony-service
CONFIG (Version=10.5)
=====================
Version 10.5
Max phoneload sccp version 17
Max dspfarm sccp version 18
Cisco Unified Communications Manager Express
For on-line documentation please see:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html
protocol mode default
ip source-address IP port 2000
ip qos dscp:
ef (the MS 6 bits, 46, in ToS, 0xB8) for media
cs3 (the MS 6 bits, 24, in ToS, 0x60) for signal
af41 (the MS 6 bits, 34, in ToS, 0x88) for video
default (the MS 6 bits, 0, in ToS, 0x0) for serviceservice directed-pickup
load 7916-24 cmterm-7916.1-0-4-2.cop.sgn
load 7965 SCCP45.9-4-2-1S
load 6921 SCCP69xx.9-2-1-0
max-ephones 35
max-dn 50
max-conferences 4 gain 6
dspfarm units 1
dspfarm transcode sessions 0
dspfarm 1 confprof1
conference hardware
privacy
no privacy-on-hold
hunt-group report delay 1 hours
hunt-group logout DND
max-redirect 10
cnf-file location: flash:
cnf-file option: PER-PHONE
network-locale[0] RU (This is the default network locale for this box)
network-locale[1] RU
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] RU (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line-mode single
moh music-on-hold.au
time-format 24
date-format dd-mm-yy
timezone 32 Russian Standard/Daylight Time
call-forward pattern .T
transfer-pattern .T
keepalive 180 auxiliary 30
timeout interdigit 10
timeout busy 10
timeout ringing 180
timeout transfer-recall 0
timeout ringin-callerid 8
timeout night-service-bell 12
caller-id name-only: enable
system message
web admin system name
web admin customer name Customer
edit DN through Web: enabled.
edit TIME through web: enabled.
background save interval 10 minutes
Log (table parameters):
max-size: 150
retain-timer: 15
create cnf-files version-stamp Jan 01 2002 00:00:00
transfer-system full-consult
transfer-digit-collect new-call
local directory service: enabled.
Extension-assigner tag-type ephone-tag.
Call List BLF is enabled
shutdown

I perform that
Router#configuration terminal
Router(config)#no logging console
Router(config)#no logging monitor
Router(config)#service timestamps debug datetime msec
Router(config)#logging buffered 40960 debugging
Router(config)#service sequence
Router(config)#no logging rate-limit
Router(config)#exit
debug tftp events
debug ephone detail
debug ephone register
In telephony-service i recreate cnf files and have this output :
Creating CNF files
IP address required is
TCP port required is 2000
read -1 bytes from flash:/its/SEPDEFAULT.cnf file
A0
0 item(s) of type 0
Unrecognized type 0 or format
Creating new SEPDEFAULT.cnf file size 58
58 bytes written OK..
ephone add http binding flash:/its/russia_lddefault.cfg failed
ephone add http binding flash:/its/russia_gkdefault.cfg failed
ephone add http binding flash:/its/russia_ffdefault.cfg failed
ephone add http binding flash:/its/united_states_lddefault.cfg failed
ephone add http binding flash:/its/united_states_gkdefault.cfg failed
ephone add http binding flash:/its/united_states_ffdefault.cfg failed
Just few seconds and i paste log
for DN 3 chan 1 to state CALL_END
009627: Mar 5 14:48:03.437: ephone-3[2/7]:UpdateCallState DN 3 chan 1 state 10 calleddn -1 chan 1
009628: Mar 5 14:48:03.437: ephone-3[2/7]:Binding ephone-3 to DN 3 chan 1 s2s:0
009629: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:Set FAC enabled (0) and dial mode (4)
009630: Mar 5 14:48:03.437: DN 3 chan 1 End Voice_Mode
009631: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009632: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone 3 incoming s2s:0
009633: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyStopMedia: normal line=1 dn=3 ch=1
009634: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyStopMedia: CloseReceive sent: normal confID=6 ref=1120
009635: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009636: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone 3 incoming s2s:0
009637: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:SkinnyStopMedia: Multimedia not active
009638: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyStopMedia: StopMedia sent: normal confID=6 ref=1120
009639: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009640: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone 3 incoming s2s:0
009641: Mar 5 14:48:03.437: ephone-3[2/7]:SpeakerPhoneOnHook
009642: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:Clean up activeline 1
009643: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:UpdateCallState unbind phone from DN 3
009644: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:ClearCallPrompt line 0 ref 0
009645: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:SkinnyCheckPendingCallBackPhone scan 2 lines
009646: Mar 5 14:48:03.437: ephone-3[2/7]:SelectPhoneSoftKeys set 0 mask FFBF for line 0 ref 0
009647: Mar 5 14:48:03.437: SkinnySetCallInfoName calling dn -1 chan 1 dn 3 chan 1,calling [] called []
009648: Mar 5 14:48:03.437: SetCallInfo DN 3 chan 1 is not skinny-to-skinny
009649: Mar 5 14:48:03.437: SkinnyStopDnRecallTimer: dn 3 chan 1
009650: Mar 5 14:48:03.437: Skinny Call State change for DN 3 chan 1 CALL_END from CONNECTED
009651: Mar 5 14:48:03.437: ephone-(3) DN 3 chan 1 calledDn -1 chan 1 callingDn -1 chan 1 :: port=0 incoming
009652: Mar 5 14:48:03.437: SkinnyUpdateCstate DN 3 chan 1 cstate 2
009653: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009654: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009655: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:SkinnyUpdateCstate first phone for DN 3 chan 1 ref 1120
009656: Mar 5 14:48:03.437: ephone-3[2/7][SEP20BBC01F9387]:UpdateCState found DN 3 on line 1
009657: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyUpdateCState process cstate 2 for inactive DN 3 chan 1 line 1 (activeLine=0 whisperLine=0)
009658: Mar 5 14:48:03.437: ephone-3[2/7]:SkinnyUpdateCstate inactive (overlay) line 1 for DN 3 ref 1120 combo=0
009659: Mar 5 14:48:03.437: DN 3 chan 1 ephone-3 state set to 2
009660: Mar 5 14:48:03.437: SkinnyGetCallState for DN 3 chan 1 IDLE
009661: Mar 5 14:48:03.437: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009662: Mar 5 14:48:03.437: ephone-3[7]:SetCallState line 1 DN 3(3) chan 1 ref 1120 TsOnHook
009663: Mar 5 14:48:03.441: ephone-3[2/7]:SkinnyTrackActiveCall for line 1 ref 1120 state 2 (slot 0)
009664: Mar 5 14:48:03.441: ephone-3[2/7]:SelectPhoneSoftKeys set 0 mask FFBF for line 1 ref 1120
009665: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009666: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009667: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009668: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009669: Mar 5 14:48:03.441: ephone-3[2/7]:Clean Up Speakerphone state
009670: Mar 5 14:48:03.441: ephone-3[2/7]:SpeakerPhoneOnHook
009671: Mar 5 14:48:03.441: ephone-3[2/7]:Speaker is not on, SpeakerPhoneOnHook suppressed
009672: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:SkinnyGetToneRef toneRef 0x0 callRef 0x460
009673: Mar 5 14:48:03.441: ephone-3[2/7]:SkinnyPhoneToneDirect: StopTone sent: normal line=1 ref=1120 tone=0x0
009674: Mar 5 14:48:03.441: Skinny StopTone sent on ephone socket [7]
009675: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009676: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009677: Mar 5 14:48:03.441: ephone-3[2/7]:SelectPhoneSoftKeys set 0 mask FFBF for line 0 ref 0
009678: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009679: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009680: Mar 5 14:48:03.441: ephone-3[7]:SetLineLamp 1 to OFF
009681: Mar 5 14:48:03.441: UnBinding ephone-3 from DN 3 chan 1
009682: Mar 5 14:48:03.441: ephone-3[2/7]:---SkinnySyncPhoneDnOverlays is onhook
009683: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009684: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009685: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:SkinnyArmPhoneCallbacks scan 2 lines
009686: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:ClearCallPrompt line 0 ref 0
009687: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:SkinnyCheckPendingCallBackPhone scan 2 lines
009688: Mar 5 14:48:03.441: SkinnyGetCallState for DN 3 chan 1 IDLE
009689: Mar 5 14:48:03.441: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009690: Mar 5 14:48:03.441: SkinnyReportDnState for overlay DN 3 chan 1 on ephone-1
009691: Mar 5 14:48:03.441: SkinnyReportDnState DN 3 chan 1 ONHOOK
009692: Mar 5 14:48:03.441: ephone-3[2/7]:SkinnyConfirmOnHookAck: dn 3 chan 1 dn_index 3 phone=2, pickupOnHook=0
009693: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:ClearCallPrompt line 0 ref 0
009694: Mar 5 14:48:03.441: ephone-3[2/7][SEP20BBC01F9387]:SkinnyCheckPendingCallBackPhone scan 2 lines
009695: Mar 5 14:48:03.445: dn_tone_control DN=3 chan 1 tonetype=0:DtSilence onoff=0 pid=418
009696: Mar 5 14:48:03.445: SkinnyGetCallState for DN 3 chan 1 IDLE
009697: Mar 5 14:48:03.445: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009698: Mar 5 14:48:03.445: ephone-3[2/7]:Check toneOn state for last_phone
009699: Mar 5 14:48:03.445: SkinnyGetCallState for DN 3 chan 1 IDLE
009700: Mar 5 14:48:03.445: called DN -1 chan 1, calling DN -1 chan 1 phone -1 incoming s2s:0
009701: Mar 5 14:48:03.665: ephone-3[2/7][SEP20BBC01F9387]:Update Stats Total for DN 3 chan 1
009702: Mar 5 14:48:03.761: ephone-3[2/7][SEP20BBC01F9387]:MediaPathEventMessage Handset OFF
009703: Mar 5 14:48:03.761: ephone-3[2/7]:MediaPathEventMessage

Help Understanding Codecs for V

My general problem is that when using Avaya VoIP client, my outgoing voice becomes garbled to the end user after a few minutes of conversation. My CPU, an Intel Pentium 4 (3GHz) also kicks into 50% utilization during a conversion, although when the conversations starts, it usually sits at about 7%. I have a high speed 5M download 52 K upload internet line. My sound card is an Audigy 2 ZS. I have eliminated obvious, like insuring other programs are not taking up CPU. I have 2 GB of high speed RAM also. I have also verified that the utilization is from the process in task manager for the IP phone client.
I am wondering since the CPU is being hammered, if I am using a software codec for the VoIP communication link. My device manager shows 3 available audio codecs. I assume that a sound card has its own embedded digital signal processing capabilities that should be capable of doing audio compression. So, my question for any hardware sound blaster guru out there is: Does the sound card do G.729 or G.7 audio compression, or is audio compression for VoIP done by a software codec?
Any comments regarding how audio codecs work in regards to VoIP will be appreciated.
Thanks,
gerryj

Fuzzy Barsik wrote:
With more than high probability common people won't be able to playback anything in MXF container.
So which would format would YOU recommend I hand them for a playable safe-keeping format? (some are on PC and some are on mac if that makes any difference)
Fuzzy Barsik wrote:
If you properly graded your 8-bit footages, i.e. in 32-bit working space (in terms of PrPro that means enabling 'Maximum Bit Depth' in Sequence Settings), 10-bit render (for which you need to check 'Maximim Bit Depth' in the Export Settings dialog, unless you render into DNxHD in MOV container) preserves quality better than 8-bit. However, if rendering to 8-bit doesn't result in colour bending or something like that, common people will hardly see any difference.
That's interesting. I've never checked that Max Bit Depth box in Sequence Settings. I don't do much grading. Color correction, yes, but nothing too crazy or with tons of layers. I figured checking that would slow things down and speed is paramount. On occasion, I'll noticed some color banding with my footage (like in the gradient of the sky) but not often and no one ever complains. I used to check Max Bit Depth in export settings not knowing what I was doing but it sounded important so I did...but then I stopped when I had THIS>> (http://forums.adobe.com/message/4773556)< issue with one of my exports (I changed many settings to get it resolved, and it only went away when max bit depth was unchecked (granted, now that you mention it, I didn't have max bit depth checked during editing in the sequence settings...hmmm).

'voice-class codec' for SCCP phones (CME)?

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