Audio drop outs on VOIP calls, but only in one direction

Posted on 2015-02-16
Last Modified: 2015-03-30
We have a local PBX server powered by Asterisk software and we periodically (3-4 times per day) experience audio drop outs, but only on the external caller's side of the conversation.  The caller can always here us but we cannot hear them.

When the drop occurs it affects all the calls that are active.  We have 10 lines connected via a T1 MIS (managed internet service) connection from AT&T.  If we have 5 concurrent calls the drop is experienced on all 5 calls.  The drop out occurs for 10-20 seconds and then the audio comes back.

I have replaced the router, core switches, tested cables, and changed VOIP providers.  Nothing has resolved the problem.  PLEASE HELP.  This has been going on for 5 months!
Question by:DarrenMcIntyre
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LVL 37

Expert Comment

by:Bing CISM / CISSP
ID: 40613440
> but only on the external caller's side of the conversation.  The caller can always here us but we cannot hear them.

just to clarify, does it mean the external caller can always hear your side when the problem happens? or in other words, the connection still keeps but becomes one way?

Author Comment

ID: 40613496
Yes, the call stays connected. The external caller can always hear us, we just cannot hear them for 10 and 20 second intervals. Then the call usually continues without any problems.  On long calls the audio drop can occur multiple times.
LVL 37

Expert Comment

by:Bing CISM / CISSP
ID: 40613556
the symptom may indicate your uplink's bandwidth is not sufficient. three options to try:

1. try changing the codec to G.729, if current one is not.

2. if some other applications or users are accessing the internet at the internet, try stopping them to see if it could help.

3. if QoS is enabled, try to disable it and see the difference. if disabled, try enabling QoS give sufficient bandwidth for SIP traffic.

moreover, check below steps if you didn't try before.

Author Comment

ID: 40621173
Thanks for the suggestions and link.  I read the recommendations and most of them I already tried.

1.  I have a T1 dedicated line just for VOIP service and we only have 10 lines, so bandwidth should not be an issue.  Also, this problem will occur even if I only have 1 external call active.
2. There are no other applications using this connection.  I have a separate connection for our internet service.
3. I have enabled and disabled QoS, STP, and SIP-ALG.  All changes had no affect.

Any other suggestions?
LVL 23

Expert Comment

by:Mohammed Hamada
ID: 40627660
What asterisk version are you using? FreePBX? what version are you on now? have you considered getting support from Asterisk forums to see if it's related to software bug ?
LVL 15

Expert Comment

ID: 40628051
Two things  --

Check our firewall, logs and see if there is anything there, also check /var/log/asterisk/full when this happens.

Second what Firewall are you using, and are you 1-1 or port mapped ?
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Author Comment

ID: 40629067
I am using Asterisk version 1.4.42
I am using Draytek 2920
I am forwarding ports 5060, 10,000-20,000 to the PBX
I haven't checked the log file yet.  I will do that now.  Anything in particular I should be looking for?
I haven't seen any reports of a bug of this nature.  I have been searching for any possible cause for a few months.
LVL 37

Expert Comment

by:Bing CISM / CISSP
ID: 40630272
> 10,000-20,000 to the PBX

why so many ports are to be forwarded? my ATA is behind my NAT firewalls and i configured nothing to get it work.

Author Comment

ID: 40630578
These are the udp ports that the VOIP provider requires to have open and forwarded.  I think it is to allow RTP voice streams to pass unhindered.
LVL 15

Expert Comment

ID: 40630629

    The ports are correct for RTP UDP only.   An your being this far back "I am using Asterisk version 1.4.42" I would also set source IPs to limit this traffic to the IP range of the provider.   On the Draytek, I have had issues in the past and suggest you be sure you are on current firmware.  

    The logs may nor may not indicate anything, but do more often than not.   Since we only have a symptom right now, I would look for anything in the time frame of the audio event, and you might need to turn up the verbosity level in your Asterisk box before you see it.

    Jeff -

Accepted Solution

DarrenMcIntyre earned 0 total points
ID: 40688320
I resolved this by manually setting the QoS values on all incoming traffic from the VOIP port to 5.  As I reviewed a wireshark capture of some voice calls I noticed that only our outgoing voice packets were getting QoS priority.

Author Closing Comment

ID: 40695573
None of the other comments led to a solution

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