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

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!
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DarrenMcIntyre
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DarrenMcIntyre
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bbaoIT ConsultantCommented:
> 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?
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DarrenMcIntyreAuthor Commented:
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.
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bbaoIT ConsultantCommented:
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.

http://www.voipmechanic.com/voip-one-way-audio.htm
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DarrenMcIntyreAuthor Commented:
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?
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Mohammed HamadaSenior IT ConsultantCommented:
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 ?
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PhonebuffCommented:
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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DarrenMcIntyreAuthor Commented:
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.
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bbaoIT ConsultantCommented:
> 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.
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DarrenMcIntyreAuthor Commented:
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.
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PhonebuffCommented:
Darren,

    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 -
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DarrenMcIntyreAuthor Commented:
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.
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DarrenMcIntyreAuthor Commented:
None of the other comments led to a solution
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