Advice on troubleshooting jabber quality during calls

[jherlitzke]

Hello,

We deployed CUCM 11.6 mid year 2017.  When we deployed it, we had a full fledge WAN with Iwan running and DMVPN.  We had 2 data centers one with a pub and sub and the other with a sub.  SIP trunks in both locations.  We have many remote offices and many remote employees.  We designed the phone system based on utilizing our current WAN and all calls come in and out of one of our two data centers.  No PSTN access in any of the remote branch offices.

Since then, our WAN has been decommissioned for cost savings reasons, as well as majority of our staff are remote now.  All offices use coax internet with a vpn tunnel back to one data center.  Remote users not using a vpn client connect back via expressway.

We have been getting lots of complaints about poor quality calls or choppy calls now using the Jabber client.  I realize that we are using the internet now and we do not have control over QOS for that.  I am moving all my primary cucm telephony services to one data center (one with the vpn termination) and all SIP trunks to that location in an effort to help.

What I am looking for... what reports may cucm have that might show some of the calls on jabber and what quality they may have been.  Or any suggested helpful reports.  Any other suggestions of what I can do to make jabber calls more solid and better quality for our users?

I have thought about doing some testing with some users and doing packet captures from their workstation and try to see what is happening during a bad call.  I plan on looking into the computer model we use and the image we have on them.  What is strange is that myself and other IT members that work both in and out of the office do not have any troubles at all.  Though we do not make nearly that many outbout pstn calls either.

Sorry long post I know, however, hoping maybe others out there are using Jabber over internet and might have some suggestions based on experience that they would be willing to share.  Thanks Everyone!

[Jaime Valencia]

You can start by asking your users to press CTRL + SHIFT + S to get the call statistic windows and see the packet loss and jitter on both ends of the call.

[jherlitzke]

I totally forgot all about that.. Excellent! Thank you for reminder...

[Jonathan Schulenberg]

Be sure to enable RTCP for audio calls under Enterprise Phone Configuratuon; it is disabled by default in CUCM.

The CMR records would also show call quality data and current Jabber releases will report if at the end of a call; however, CUBE continues to be a problem here because it won’t upload the other half of the CMR at the end of a call. MGCP is the only trunk protocol that does this currently.

You may want to consider placing a NAM or similar on the inside of Expressway-C so it can see the RTP traffic once it lands in the DC.

Also make sure you have good monitoring of the internet pipe at the DC. Congestion at the head-end will hurt everyone. You could also consider giving Expressway-E egress priority if your internet edge switch supports it.

[GrumbleShark]

The lack of QOS over the Internet may be the final takeaway.  If you're looking for some tools to give you MOS then you may want to look at:

Solarwinds CDR Tracker - https://www.solarwinds.com/free-tools/call-detail-record-tracker

VoIP Detective call accounting - https://www.ucguru.com/voip-detective/

Both of which are free, and should read the CMR files produced by CUCM.

[Dennis Mink]

IMHO, you can spend a lot of time quantifying call quality across the non-Qos-ed internet, but the reality is, there are no guarantees on how a call gets perceived by a user. which means one month your call detail records might be fine and the next month you have a multitude of call quality problems. the reality is that you can put all sorts of compression and low bandwidth savings in place, but it doesn't change the fundamental nature of internet, which is there are no guarantees. Tell your boss you cant have a champagne dinner on a fish and chips budget.

[jherlitzke]

Champagne dinner on a fish and ships budget.  I love that!  So true as well...

[jherlitzke]

So far I appreciate all the responses, and they have all been great.  Thank you!  Still working on it and doing some testing, but in the end it really does come down to all calls are using the internet and there is no way to control that.  I miss our WAN...