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Location CAC not working on SIP trunk

voiphub01
Level 1
Level 1

Hi,

Below is the call flow;

Phone (CAC location NY) -> CUCM -> SIP Trunk (CAC location SJ) -> Gateway

I want to allow only two calls (G729) between location NY and SJ hence created two locations in CUCM as NY and SJ. Bandwidth between these two location is configured as 50. Regions are created and assigned for calls to use G729 codec. Location is assigned to Phone and SIP Trunk directly.

Now the issue is system is allowing more than two calls on SIP trunk.

I verified the same in serviceability and checked the effective path. For first call, I can see the location reduced from 50 (-24) to 26. Second call, it further reduced from 26 (-24) to 2. Now when there is a third call, system allows it and available bandwidth reflects as 0. User can also make further calls now.

I tried to find out service parameter which can control this behavior but failed.

Regards 

1 Accepted Solution

Accepted Solutions

Hi,

It seems to be related to the following

https://tools.cisco.com/bugsearch/bug/CSCum41473/?reffering_site=dumpcr

I believe this is fixed in CUCM 11.0 with the introduction of the following service parameter:


Deduct Audio Bandwidth from Audio Pool for Video Calls

Use this procedure if you want to split the audio and video bandwidth deductions into separate pools for video calls. By default, the system deducts the bandwidth requirement for both the audio stream and video stream from the video pool for video calls.

When you enable this feature, CAC includes the bandwidth required for the IP/UDP network overhead in the audio bandwidth deduction. This audio bandwidth deduction equates to the audio bit rate plus the IP/UDP network overhead bandwidth requirement. The video bandwidth deduction is the video bit rate only.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide/CUCM_BK_C733E983_00_cucm-system-configuration-guide-transformed_chapter_0101011.html#CUCM_TK_DB12677A_00

Manish

View solution in original post

8 Replies 8

voiphub01
Level 1
Level 1

An update:

Just found that issue is only with devices which supports video e.g., 8945 etc. Other non video phones works as expected means drop the third call. If I disable video/camera on device page, 8945 also started working as expected.

Audio bandwidth between NY and SJ is 50, and video is default 384. Since it's a gateway call, although call is being made from video phone but it's not a video call. Seems like even audio call from video device is utilizing video bandwidth. But serviceability is not reducing any bandwidth from video available bandwidth. Not sure what is happening from video phone.

Regards 

Hi,

As per the SRND

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/cac.html#pgfId-1489908

Table 13-5 Bandwidth Pool Usage per SIP Trunk and Endpoint Type

Endpoint
SIP Trunk
Locations and Links Pool Used

TelePresence endpoint

Immersive

Immersive bandwidth

TelePresence endpoint

Desktop

Immersive and video bandwidth

TelePresence endpoint

Mixed

Immersive and video bandwidth

Desktop endpoint

Immersive

Immersive and video bandwidth

Desktop endpoint

Desktop

Video bandwidth

Desktop endpoint

Mixed

Immersive and video bandwidth

Non-video endpoint

Any

Audio bandwidth

By default, all video calls from either immersive or desktop endpoints are deducted from the locations and links video bandwidth pool. To change this behavior, set Unified CM’s CallManager service parameter Use Video BandwidthPool for Immersive Video Calls to False, and this will enable the immersive video bandwidth deductions. After this is enabled, immersive and desktop video calls will be deducted out of their respective pools.

Manish

Appreciate your response.

I have changed this service parameter to True however there is no change in the behavior. Although audio bandwidth between these two locations is set to 50, I can still make more that two G.729 calls. Effective audio bandwidth after two calls shows None in serviceability but it is still allowing more than two calls. Moreoever, no deduction in video and immersive bandwidth is being shown, shows default 384 even after two calls.

My requirement is to allow only 2 G.729 calls from SIP trunk. What configuration changes I need to have there?

Regards

Can you share screenshot of the region and location config page for these two locations.

Manish

1. Audio bandwidth set to 8 Kbps between NewYork and SanJose. Video bandwidth is default.

re

2. Location audio bandwidth is set to 50. Video is again default.

lo

3. Phones are in Newyork region and location. SIP trunk is in SanJose region and location. Where there is one call on SIP trunk, we can see the available bandwidth;

one

4. When there are two calls, we can see available bandwidth as 2.

two

5. Now SIP trunk should not allow third call, but it's not restricting.

th

Hi,

It seems to be related to the following

https://tools.cisco.com/bugsearch/bug/CSCum41473/?reffering_site=dumpcr

I believe this is fixed in CUCM 11.0 with the introduction of the following service parameter:


Deduct Audio Bandwidth from Audio Pool for Video Calls

Use this procedure if you want to split the audio and video bandwidth deductions into separate pools for video calls. By default, the system deducts the bandwidth requirement for both the audio stream and video stream from the video pool for video calls.

When you enable this feature, CAC includes the bandwidth required for the IP/UDP network overhead in the audio bandwidth deduction. This audio bandwidth deduction equates to the audio bit rate plus the IP/UDP network overhead bandwidth requirement. The video bandwidth deduction is the video bit rate only.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_0_1/sysConfig/CUCM_BK_C733E983_00_cucm-system-configuration-guide/CUCM_BK_C733E983_00_cucm-system-configuration-guide-transformed_chapter_0101011.html#CUCM_TK_DB12677A_00

Manish

Yes, seems what I am facing. But strange until version 11, there is no solution. So how customers are managing to use location for audio calls (from video terminal) on SIP trunk before v11?

Regards

I don't see any workaround for this in 10.x or lower versions.

Manish