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New Member

Call Admission Control implementation on CUCM8.5

Hello to all!

     I have one question about Call Admission Control and it is the following: I understasnd the fact of configuring the maximum bandwidth for audio between Locations and setting the codec relationships between Regions, but wich codec is used in the Call Leg established between the Voice Gateway and the CUCM. Let's assume that the Voice Gateway is configured as an h323 gateway on the CUCM. Lets say I configured codec G711 for that outbound dial peer in the voice gateway, but all the regions are configured to use G729 between them and in themselves, will it work or should ther codec on the outbound dial peer between the Voice gateway and the CUCM needs to be changed to G729? Thanks in advancved guys! 

1 ACCEPTED SOLUTION

Accepted Solutions
Green

Call Admission Control implementation on CUCM8.5

Hi,

So it sounds like your PSTN access is via an SP using SIP is that correct ?

You should be able to run 20 session on your PVDM3-64

Try adding this to yor gateway,

Correct the IP address and interface numbers to suit your set up

Also correct the  SCCP version number to as high and as near to CUCM 8.5 as you can

Issue the command like this

identifier 1 priority 1 version ? to see the available versions

!

voice-card 0

dsp services dspfarm

!

!

sccp local GigabitEthernet0/0

sccp ccm 192.168.70.2

identifier 1 priority 1 version 5.0.1

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register XCODER1

!

dspfarm profile 2 transcode XCODER1

codec g729r8

codec g729br8

codec g711ulaw

codec g711alaw

codec g722-64

maximum sessions 20

associate application SCCP

!

HTH

Alex

Regards, Alex. Please rate useful posts.
10 REPLIES
Green

Call Admission Control implementation on CUCM8.5

Hi,

On the CUCM your gateway will use a devic pool DP.
The DP will have region set.
The region will use different codecs to oother regions.

Lets say you have 3 regions

1) HQ
2) BRANCH-A
3) BRANCH-B

Region HQ uses G711 to HQ
Region HQ uses G711 to BRANCH-A
Region HQ uses G729 to BRANCH-B

Region BRANCH-A uses G711 BRANCH-A
Region BRANCH-A uses G711 to HQ
Region BRANCH-A uses G729 to BRANCH-B

Region BRANCH-B uses G711 BRANCH-B
Region BRANCH-B uses G729 to HQ
Region BRANCH-B uses G729 to BRANCH-A

If a PSTN call comes in on the Voice Gateway at BRANCB-B the
the Gateway should be in REGION BRANCH-B. If the call is aimed at a phone in BRANCH-B
then the codec should be negotiated between the Gateway and the CUCM to use G711

If the call is aimed to say BRANCH-A then the negotiation should be to G729

This negotiation should take place under H323 umbrella using
the H.245 capability exchange

To configure the Gateway something like this:-

!
! This part ensure timely H323 TCP setups on port TCP 1720
!
voice class h323 1
h225 timeout tcp establish 3
!
!
!This part allows for codec negotiation
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
!
dial-peer voice 11 voip
description *** INCOMING to CUCM SUB1 **
preference 1
destination-pattern 701....
incoming called-number .
voice-class codec 1
voice-class h323 1
session target ipv4:10.10.10.12
dtmf-relay h245-alphanumeric
ip qos dscp ef media
ip qos dscp cs3 sig
no vad
!

HTH

Alex

Regards, Alex. Please rate useful posts.
New Member

Call Admission Control implementation on CUCM8.5

Hello Alex and thanks for your reply! I have four regions configured as follows in the CUCM 8.5:

This is from the perspective of region RG_GCHC wich is the main site:

RG_GCHC  64 kbps (G.722, G.711) 

RG_San_Lorenzo  8 kbps (G.729) 

RG_Trujillo_Alto  8 kbps (G.729) 

RG_SiVif  8 kbps (G.729)

From the perspective of region RG_SiVif:

RG_GCHC  8 kbps (G.729) 

RG_San_Lorenzo  8 kbps (G.729)   

RG_Trujillo_Alto  8 kbps (G.729)   

RG_SiVif  64 kbps (G.722, G.711) 

From the perspective of region RG_San_Lorenzo:

RG_GCHC  8 kbps (G.729)   

RG_San_Lorenzo  64 kbps (G.722, G.711)  

RG_SiVif  8 kbps (G.729)   

RG_Trujillo_Alto  8 kbps (G.729)

RG_GCHC  8 kbps (G.729) 

RG_San_Lorenzo  8 kbps (G.729)

RG_SiVif  8 kbps (G.729) 

RG_Trujillo_Alto  64 kbps (G.722, G.711)

RG_GCHC  8 kbps (G.729)   

From the perspective of region RG_Trujillo_Alto:

RG_GCHC  8 kbps (G.729) 
RG_San_Lorenzo  8 kbps (G.729)
RG_SiVif  8 kbps (G.729) 
RG_Trujillo_Alto  64 kbps (G.722, G.711) 

There is only one Voice Gateway and it is in the main site and has region RG_GCHC configured. All incoming calls are answered in the main site by either a directory number or the AA.

I configured the voice class parameters in the Voice GW and the outbound dial-peer to the CUCM8.5 as follows:

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

!

voice class h323 1

  h225 timeout tcp establish 3

dial-peer voice 200 voip

translation-profile outgoing GCHC_Main_Pilot

preference 1

destination-pattern 787687....

session target ipv4:192.168.70.2

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

ip qos dscp cs3 signaling

no vad

After this configs, if there is a call from the PSTN to the main reg, it gets answered but if its transfered to another extension on one of either of the other sites, as soon as the directory number on the other region answrers the transfered call from the PSTN, the calls get dropped. I know is because of mismatch of codecs but I still cant figure it out. Please help and thanks again!

Green

Call Admission Control implementation on CUCM8.5

Hi,

For your scenario of calls being transferred from G711 to G729 regions you will need a transcoder.

Have you already registered a IOS transcoder with the CUCM.
It will need to be put in a MRGL that the gateway has access to.

Regards
Alex

Regards, Alex. Please rate useful posts.
New Member

Call Admission Control implementation on CUCM8.5

No, I havent registered an IOS transcoder with the CUCM. How can I achieve this? It is a new hardware needed?

Green

Call Admission Control implementation on CUCM8.5

Hi,

A transcoder uses DSP resources from voice gateway type routers.
These can be on the same physical router as say E1/T1 Pri , BRI or FXO etc.
You just have to have spare sufficient DSP resources.

You also can use a dedicated router just puely for Media resources like
Transcoders, MTP and Hardware conferencing.

There are many documents available to read about this.

For CUCM 8.5
SRND
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/media.html#wp1046264

Administration
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmcfg/b04trans.html

General
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008084fe1f.shtml

In the interim,
You could try setting up a new region with G729 to all other regions
Set up a new Location with a bandwidth of 24K x the number of simultaneous calls you expect onm the gateway,
E.g.
If you reckon that the router will max at 20 simultaeous calls then create th new location
to be 24K x 20  = 480K

Create a new device pool and use the new location & region in it.
Then change the H323 gateway to use the new location

Then change the IOS dial peer to :-

!
!
dial-peer voice 200 voip
translation-profile outgoing GCHC_Main_Pilot
preference 1
destination-pattern 787687....
session target ipv4:192.168.70.2
no voice-class codec 1
codec g729r8
voice-class h323 1
dtmf-relay h245-alphanumeric
ip qos dscp cs3 signaling
no vad
!

Now all the calls to and from the gateway should be G729 and should be able to
transfer to all regions

HTH
Alex

Regards, Alex. Please rate useful posts.
New Member

Call Admission Control implementation on CUCM8.5

Excellent Alex! After reading some documentation about the transcoding configuration on voive gateway, I have the following doubts:

1. I have a Cisco ISR 2921 with PVDM3-64 as the voice gateway and will use it as the Transcoder. How can I determine the maximum number of sessions for the dsp farm profile configuration for transcoding?

2. Can I only configure the transcoding dsp farm profile or do I need to configure also the profiles for conference?

Thanks again!

New Member

Call Admission Control implementation on CUCM8.5

I tried the procedure you mentioned above but still doesnt work for calls from the PSTN bcause the provider is using g711.

Green

Call Admission Control implementation on CUCM8.5

Hi,

So it sounds like your PSTN access is via an SP using SIP is that correct ?

You should be able to run 20 session on your PVDM3-64

Try adding this to yor gateway,

Correct the IP address and interface numbers to suit your set up

Also correct the  SCCP version number to as high and as near to CUCM 8.5 as you can

Issue the command like this

identifier 1 priority 1 version ? to see the available versions

!

voice-card 0

dsp services dspfarm

!

!

sccp local GigabitEthernet0/0

sccp ccm 192.168.70.2

identifier 1 priority 1 version 5.0.1

sccp

!

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register XCODER1

!

dspfarm profile 2 transcode XCODER1

codec g729r8

codec g729br8

codec g711ulaw

codec g711alaw

codec g722-64

maximum sessions 20

associate application SCCP

!

HTH

Alex

Regards, Alex. Please rate useful posts.
New Member

Call Admission Control implementation on CUCM8.5

Hi Alex!

     Yes, my PSTN up-link is a SIP trunk to the SP SIP server. Thanks Alex, it seems to be working now. I changed the Regions as follows:

From the perspective of region RG_GCHC wich is the main site:

RG_GCHC  64 kbps (G.722, G.711) 

RG_San_Lorenzo  8 kbps (G.729) 

RG_Trujillo_Alto  8 kbps (G.729) 

RG_SiVif  8 kbps (G.729)


From the perspective of region RG_SiVif:

RG_GCHC  8 kbps (G.729) 

RG_San_Lorenzo  8 kbps (G.729)   

RG_Trujillo_Alto  8 kbps (G.729)   

RG_SiVif  64 kbps (G.722, G.711) 


From the perspective of region RG_San_Lorenzo:

RG_GCHC  8 kbps (G.729)   

RG_San_Lorenzo  64 kbps (G.722, G.711)  

RG_SiVif  8 kbps (G.729)   

RG_Trujillo_Alto  8 kbps (G.729)


From the perspective of region RG_Trujillo_Alto:

RG_GCHC  8 kbps (G.729) 

RG_San_Lorenzo  8 kbps (G.729)

RG_SiVif  8 kbps (G.729) 
RG_Trujillo_Alto  64 kbps (G.722, G.711) 

One last question, how is the maximum nuber of sessions calculated for the PVDM3-64? Thanks again for your support. Great work!

Green

Call Admission Control implementation on CUCM8.5

Hi,

You need to use the DSP calculator

http://www.cisco.com/web/applicat/dsprecal/dsp_calc.html

Glad I was able to help you to fix the issue

Regards

Alex

Regards, Alex. Please rate useful posts.
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