cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
Announcements
393
Views
0
Helpful
1
Replies
Highlighted
Beginner

UCC | CUCM 11.5 | Voice and Video | WebEx Services | Jabber | Telepresence | bandwidth and configuration

Hi everybody,

 

I am new on the UCC environment, normally I am working as Junior Network Administrator.

But at the end I need to run and manage this environment and need help by some questions.

 

We're facing a lot of quality issues in the Jabber as Softphone and Telepresence environment.

Only one-direction audio, robotic-Voice, mosaic pictures/video.

We've configured the Telepresence as SIP-Devices and deactivated H.323 and also set the DSCP values from our WAN-Provider for the Telepresence Systems.

Separate VLAN for the Telepresence Systems also tagged as Voice in the L2 Segment.

The TMS is just for the phonebook.

 

As far as I Know we use for Jabber the OPUS Codec by default, but where can I check this settings/values ?

Is there something like the DSCP configuration needed to set up for the Jabber Client, if yes where can we do it?

Firewall-Rules are set as recommended/required from Cisco.

 

 

 

How many bandwidth I need for:

  • a classical voice Jabber call.
  • a video Jabber call.
  • a classical 1:1 Telepresence session.
  • a multiple Telepresence session.
  • If a user wants to join a WebEx Meetings. d ( is there a difference in the DSCP values if it is just audio or Video)?
  • If we've a WebEx Teams call only with voice.
  • If we've a WebEx Teams call with video.
  • If we've a WebEx Teams video session in a space.

 

What is needed to know in this case:

  • What do I need to know in the CUCM in his settings?
  • What do I need to know in the TMS in his settings?
  • What do I need to know in the CMS in his settings?
  • What do I need to know in the Expressway-C and Expressway-E in his settings? 

 

I hoppe you can help me the answer this questions and get a better understanding for this set-up.

 

1 REPLY 1
Highlighted
Beginner

Not a comprehensive answer by any means, but I would start looking at lower layer issues first, and work your way up, as in:

 

Where are your folks experiencing Jabber voice quality issues, over wifi, VPN/Expressway, etc? Can you localize your issues to a certain segment of your network?

 

Also, can you ensure that DSCP markings are being applied by all applications/appliances in your communications path? If the edge switch is not marking packets on the return trip, or you are not establishing QoS for your other applications and giving the real time audio, expedited forwarding won't be honored, and you might experience the issues you are describing.

 

As far as the one way audio issues, in my experience, that usually is a codec missmatch between the source and destination streams, or could simply be a FW port issue, especially if your rules aren't stateful of traffic streams.

 

I would also see what utilization levels I expect, how many users do I have, how often do they make UC calls, and to where(intranet vs. PSTN) and when(peak times). This allows you to establish a baseline with which you can better plan and support the load your users exercise on the system.

 

For CUCM specifics, I would look at the below documentation for guidance:

CUCM and QoS Recommendations:

https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/43587-ccm-and-qos.html

 

Jabber Deployment Pre-Check and Considerations:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/10_5/CJAB_BK_D6497E98_00_deployment-installation-guide-ciscojabber/CJAB_BK_D6497E98_00_deployment-and-installation-guide-for_chapter_00.html

 

Cisco Webex Guide:

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/CWMS/b_administrationGuide/b_administrationGuide_chapter_01101.html

Content for Community-Ad