Solved

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

Posted on 2015-02-16
13
220 Views
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!
0
Comment
Question by:DarrenMcIntyre
[X]
Welcome to Experts Exchange

Add your voice to the tech community where 5M+ people just like you are talking about what matters.

  • Help others & share knowledge
  • Earn cash & points
  • Learn & ask questions
  • 6
  • 3
  • 2
  • +1
13 Comments
 
LVL 37

Expert Comment

by:bbao
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?
0
 

Author Comment

by:DarrenMcIntyre
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.
0
 
LVL 37

Expert Comment

by:bbao
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.

http://www.voipmechanic.com/voip-one-way-audio.htm
0
Supports up to 4K resolution!

The VS192 2-Port 4K DisplayPort Splitter is perfect for anyone who needs to send one source of DisplayPort high definition video to two or four DisplayPort displays. The VS192 can split and also expand DisplayPort audio/video signal on two or four DisplayPort monitors.

 

Author Comment

by:DarrenMcIntyre
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?
0
 
LVL 24

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 ?
0
 
LVL 15

Expert Comment

by:Phonebuff
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 ?
0
 

Author Comment

by:DarrenMcIntyre
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.
0
 
LVL 37

Expert Comment

by:bbao
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.
0
 

Author Comment

by:DarrenMcIntyre
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.
0
 
LVL 15

Expert Comment

by:Phonebuff
ID: 40630629
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 -
0
 

Accepted Solution

by:
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.
0
 

Author Closing Comment

by:DarrenMcIntyre
ID: 40695573
None of the other comments led to a solution
0

Featured Post

[Live Webinar] The Cloud Skills Gap

As Cloud technologies come of age, business leaders grapple with the impact it has on their team's skills and the gap associated with the use of a cloud platform.

Join experts from 451 Research and Concerto Cloud Services on July 27th where we will examine fact and fiction.

Question has a verified solution.

If you are experiencing a similar issue, please ask a related question

Let’s list some of the technologies that enable smooth teleworking. 
If your business is like most, chances are you still need to maintain a fax infrastructure for your staff. It’s hard to believe that a communication technology that was thriving in the mid-80s could still be an essential part of your team’s modern I…
Sending a Secure fax is easy with eFax Corporate (http://www.enterprise.efax.com). First, just open a new email message. In the To field, type your recipient's fax number @efaxsend.com. You can even send a secure international fax — just include t…
Monitoring a network: how to monitor network services and why? Michael Kulchisky, MCSE, MCSA, MCP, VTSP, VSP, CCSP outlines the philosophy behind service monitoring and why a handshake validation is critical in network monitoring. Software utilized …
Suggested Courses

626 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question