cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2785
Views
10
Helpful
7
Replies

Jabber - One way speech for few seconds (No MRA)

J_Cormier
Level 1
Level 1

Hi,

We have users using Jabber with Cisco VPN and some users have one way speech since few weeks. The only changes we can see are windows updates.

  • Packet loss shown on jabber statistic window.
  • No packet loss shown on Cisco ASA connection.
  • Sip inspect has been disabled.
  • Occurred on Jabber to Jabber and Sip gateway to Jabber.
  • MARI has been disabled.
  • We are not using MRA.
  • If we disconnect the headset and reconnect or change the audio settings, the audio start working.
  • Occurred with hands-free and with usb or jack headset.
  • Jabber version 12.8, 12.9 and 14 tested.
  • In the wireshark trace, we have out-of-order / wrong sequence packets. and seem to have duplicate packets.
  • Occur Cabled and on wifi.
  • We had the same issue with CIPC on the agent side.
  • If the called side use CIPC, we get the audio.
  • Recording server get the audio correctly (Maybe the system use reassembly process).
  • Recording has been disabled
  • All clients with Windows 10 1909 or 20h2
  • No cpu issue (recent pc)
  • CUCM 11.8 (SU8) and IMP (SU8)
  • All update has been done on the lenovo pc
  • Occur for 1 day and day after it's working well (tested on both ASA cluster, same issue) we can switch the asa cluster (So ip address change) and we still have the issue.
  • Only few users have the issue (5/10 every day) on a total of 800 agent.
  • TAC is unable to find the issue.

Thank you for your help

7 Replies 7

Scott Leport
Level 7
Level 7

Hi,

 

One way voice issues are typically IP routing, Firewall or NAT related, but I do have a few queries which I hope can help narrow in on the issue somewhat.

 

Does the issue occur only with VPN users or is it the same with users on the LAN? Is it specifically inbound / outbound calls to / from the PSTN to the VPN users? 

You refer to some users who have issues with one way voice. Does that mean that other VPN users don't have issues with one way voice?

What's different about the affected users vs non-affected users?

Do they obtain an IP address from the same pool of addresses as your other VPN users or are they on a different pool of IP addresses?

Are the non-affected and affected users have VPNs established on the same ASA?

For outbound calls which target the SIP gateway (assuming CUBE) is the ASA your Internet gateway and therefore upstream of the CUBE?

Was SIP Inspection disabled already or just disabled for troubleshooting purposes?

Is your ASA configured to allow the RTP port range your CUBE operates on? Do a "show voip rtp connections" on your CUBE to find out the default range it operates on and ensure your ASA is allowing that entire range.

I am a little unclear on what you mean by agent.

I would also advise you take debug ccsip messages of a working call and one of a call where the issue was experienced, compare them and check what's different, e.g. different SIP signalling sources, RTP IP addresses etc.

I would also advise packet captures taken from the ASA and the CUBE if we're troubleshooting one way voice issues to / from the PSTN.

 

A topology diagram of your setup may be helpful too just in case that reveals anything, but otherwise these are some of my suggestions to start with.

 

Also, I am guessing you've seen this link too?
https://community.cisco.com/t5/collaboration-voice-and-video/how-to-troubleshoot-one-way-no-audio-issues/ta-p/3164442

 

 

 

Hi, 

Problem seem to be with windows defender.

When the user have the issue, we update defender and it start working just after the update.

We are adding the exclusion to the AV to see if it help.

 

Thank you

Hello,

 

we're experiencing the same issue with Jabber Rel. 14 on some devices with Windows Defender enabled.

Could you advise me about the Windows Defender Version you installed or any Settings you changed to fix the issue.

 

Thanks in advance for your reply.

BR,
Andrea

Hi Andrea,

We just added exclusions. We summited it to Cisco and there is the bug

https://bst.cisco.com/bugsearch/bug/CSCwa76267

But note we rolled back from 14.0.2 to 12.8.6 because we had other issues with 14.0.x and 12.9.6 (Hold\Resume issue and headset hold notification issue))

Now we have a bug with 2 audio stream sent to the client on outbound calls causing bad quality audio and out of order.

Hope it help.

J_Cormier
Level 1
Level 1

Thank you for your time and sorry for the delay. 

 

Q.Does the issue occur only with VPN users or is it the same with users on the LAN? Is it specifically inbound / outbound calls to / from the PSTN to the VPN users?
A.Jabber to Jabber internal calls and inbound call from sip trunk. We dont have call center user in the office, they are all vpn, but no case with physical phone at the office.

 

Q.You refer to some users who have issues with one way voice. Does that mean that other VPN users don't have issues with one way voice?
A.Some users have the issue at the morning, and the issue disappear in the afternoon but appear for other user in the afternoon. No ip address changes seen and dont see asa fallback.

 

Q.What's different about the affected users vs non-affected users?
A.We dont dont yet All brand new lenovo pc. We may have more issue with pc than laptop, but not really sure about that. We tested with a different user logged to the pc and we have the same issue.

 

Q.Do they obtain an IP address from the same pool of addresses as your other VPN users or are they on a different pool of IP addresses?
A.We cannot find a ip address range having more issue or they dont have more problem on 1 ASA cluster than the other.

 

Q. Are the non-affected and affected users have VPNs established on the same ASA?
A. Yes on the same ASA

 

Q. For outbound calls which target the SIP gateway (assuming CUBE) is the ASA your Internet gateway and therefore upstream of the CUBE?
A. We have the issue within the internal network so I will discard the Sip gateway/Cube

 

Q. Was SIP Inspection disabled already or just disabled for troubleshooting purposes?
A. Was enabled on 2/3 asa. Now 3/3 are disabled.

Q.Is your ASA configured to allow the RTP port range your CUBE operates on? Do a "show voip rtp connections" on your CUBE to find out the default range it operates on and ensure your ASA is allowing that entire range.
A.I will need to check the ASA firewall rule, but it's mostly any/any within the internal network/vpn

 

Q.I am a little unclear on what you mean by agent.
A.Agent is user logged into Finesse/ICM

 

Q.I would also advise you take debug ccsip messages of a working call and one of a call where the issue was experienced, compare them and check what's different, e.g. different SIP signalling sources, RTP IP addresses etc.
A.We have the same issue in nternal network, we looked to SDL trace and cannot find any informations.

 

Q.I would also advise packet captures taken from the ASA and the CUBE if we're troubleshooting one way voice issues to / from the PSTN.
A.We take capture from 2 PC with Jabber, the sender see packet loss, but in the trace we can see duplicate packet.1 with fffffff payload (Silence) and another with normal payload. so it give us Out-of-order/wrong sequence packet

 

Note : We also installed CIPC on user's pc with the issue and we got the same behavior, but if the called number have cipc it's working fine.

Note : If we enable MTP, the audio is working but it's really bad

 

Hi,

 

Regarding the internal calls issue, are the Jabber devices registered to the same CUCM node? Do you have another CUCM node you can move the Jabber CSF devices to? If so, does the problem remain when registered to another CUCM node?

 

When you were checking the RTP statistics in Jabber, were the Tx and Rx stats increasing together when the issue occurred or was one increasing and not the other?

 

If nothing is gained by moving the Jabber CSF devices to another CUCM node, then look at IP routing for any internal routing issues which correlate with the time of day the issue occurs and make sure the Firewalls are allowing the appropriate RTP port ranges between all your necessary networks. 

 

Key points to take from the link supplied in my first post:

1. If the phone is transmitting, but the other side is not receiving, it's a network issue.

2. If you don't see the phone transmitting, get the CCM traces for the phone call and check if the phone is receiving a send only / receive only SDP.

We tried to change the cucm/device pool without success.

In fact we don't see packet loss (CTRL+Shift+S shown packet loss, but the rtcp is bad

because of the duplicate packets I think)

No PL at the user workstation,

No PL at the asa connexion

No PL for the caller

 

So we now tried to find if it's cause by something on the pc.

 

Thank you