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8831 one way audio

Scuromano
Level 1
Level 1

Several 8831s at a site report one way audio periodically.  The users dial into an external, third party conferencing service.  Once in a while, everyone else can't hear the users on the 8831.  They have to hang up and dial in again.  The firmware is sip8831.10-3-1SR4-2-NA, CUCM version 10.5.1.  The problem will, of course, disapear for a week or so and come back.  There are 10 total 8831s at the site, but only 4 complain.  There are around 100 8831 phones in the whole company and only one phone in one other site has complained of the exact thing.  Always the other callers cannot hear the users on the 8831.  They say, when they call the conference service from their desk phones 8841s, there is no problem.

Anything else to look at?

16 Replies 16

Dennis Mink
VIP Alumni
VIP Alumni

you would need to have that user contact tyou after the issue has occcured so you can use RTMT and pull the call traces. typically one way audio is a routing issue or and access-list blocking rtp in the path.

 

without addidtional information its anyone's guess what the cause is.

Please remember to rate useful posts, by clicking on the stars below.

Daniel Gagne
Level 1
Level 1

Hi,

 

Have you upgraded to sip8831.10-3-1SR4-2 recently?

 

If so, you have hit the same issue that we encountered here.

 

There seems to be a bug with this firmware version and "older" 8831 in the field.

Here we have 96 8831 and we received a new batch requiring at least firmware version 10-3-1-SR3 and these don't have the issue with the new firmrare....only the previous 8831 have the issue.

 

According to the diagnostics we did, the issue is with the full duplex, as soon as both parties try to talk at the same time, the 8831 users can't be heard by the others. They have to wait for a few seconds without talking for the voice from the 8831 to be heard again.

 

Look at the release notes and you'll see a bug for that :

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/8831/firmware/10-3-1-sr4/cs38_b_rns-8831-8831nr-1031sr4.html

 

The bug id is: CSCvh12266 One-way audio during full duplex streaming from 8831

 

I have a TAC case about this but the engineer can't confirm that we are hitting this bug and won't tell me why we can't access the détails on the bug search tool.

 

Anyway, TAC gave us access to firmware version sip8831.10-3-1ES9-1 and we are currently testing it on two of the affected devices , the ones used by the persons that have reported the issue.

 

All the other 8831 have been downgraded to version sip8831.10-3-1SR2-2.

 

By the way, the newer version of 8831 we received can't go back to sip8831.10-3-1SR2-2, they just stay at the latest version or the version that they came with.

 

You can open a TAC case and request access to the ES firmware above or just rollback to the prior version of the firmware, through CUCM's Device Defaults page.

 

 

 

 

 

Hi Daniel,

 

Thanks for the post same issue here.

 

Cannot downgrade and stuck with the one way audio issue. Contacted the TAC who have informed us they do not have access to sip8831.10-3-1ES9-1

 

Very frustrating considering we did a bug scrub using the bug tool kit prior to choosing this load.

Downgrade bug - CSCvf75169

This is good information.  This could be the issue.  There are over 50 8831s in the company but only a couple phones, in two sites are complaining about this.  Seems to only happen when they join a conference on the 8831s, doesn't happen when anyone uses their desk phones.  Did you say that your new phones on 10-3-1SR3 did not have the issue?  Because I have sip8831.10-3-1SR3-5 on the CUCM.

Hi Scuromano,

 

The one running 10-3-1SR3-5 didn't have the issue at first, just after upgrading to 1-3-1-SR4-2.

When we downgraded them back to their original firmware, the issue when away.

 

We still had big issues with one of the units but doing a factory reset on it cured the problem, perhaps doing a factory reset on your affected units would help.

 

Please note that 10-3-1-Es9-1, available through TAC,  also fixed our issue so we will stay on the lookout for the next official release of the firmware.

 

 

 

  I downgraded a couple phones to sip8831.10-3-1SR3-5 a few days ago.  We'll need a week or so to monitor as the customer says someitmes it goes a week without the issue.   I'll keep you updated.

 

Just a side note:  one users stated that the problem occurred after hitting the Mute button.  I guess we'll see if this fixes it.

Hello everyone,

 

Just to confirm that firmware 10-3-1-ES9-1 fix the the oneway issue on our somewhat "older" units.

 

There was only one unit on which the new firmware didn't fix the issue. We ended up doing a factory reset and then the issue was gone.

 

Looks like the firmware upgrade didn't went well on this unit.

 

Good luck to you all.

 

Yes all of our handsets required a "hard reset" after the upgrade to resolve the half duplex issue. TAC advised we disable the g722 codec and shut/no shut the port. But after further testing it appears that only the "hard reset" was required. This is the reset that requires unplugging each unit.

 

https://www.cisco.com/c/en/us/support/docs/collaboration-endpoints/unified-ip-conference-phone-8831/119143-technote-ip-phone-00.html

Hi Daniel - have you had the issue re-occur? We are currently running the ES code and the issue is re-occurring it appears to resolve itself temporarily by rebooting the phone. I have also been advised that the behavior is as expected by the BU....... I would be interested to hear other peoples feedback.

 

 Normal double talk requires the volumes on both sides are similar. 
 If one side (side A) talks in high volume, dominating the conversation, the remote side (side B)'s speaker gets loud and the microphones has high echo.
If the echo on side B is much higher than Side B party's talking voice,  Side B's AEC has to attenuate its sending volume,  which leads to the voice choppiness.

Hi,
We have been made aware that the issue is reoccurring and I've asked for the unit to be sent to me for further testing.
Unfortunately, I've not been able to reproduce the issue.

I have replicated the issue a reboot seems to resolve the issue temporarily. I have heard word that the BU is working on another ES I will keep you updated.

Usually one way audio means there's one way IP communication, remember that after call setup phones the RTP media is exchanged between phones.

You mentioned they are dialing to an external conference number, I'll check the path and the IP communication between the devices along the way (cucm, cube, sip itsp if applicable).

 

Other reason may be lack of DSP resource in case you are transcoding codes/protocol.

 

Rolando A. Valenzuela.

Issue with the code confirmed by the TAC.

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