Solved

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

Posted on 2015-02-16
13
203 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
  • 6
  • 3
  • 2
  • +1
13 Comments
 
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?
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: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.

http://www.voipmechanic.com/voip-one-way-audio.htm
0
 

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 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 ?
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
New! My Passport Wireless Pro Wi-Fi Mobile Storage

Portable wireless storage to offload, edit, and stream anywhere.

High-capacity, wireless mobile storage designed to accompany professional photographers and videographers in the field to easily offload, edit and stream captured photos and high-definition videos.

 

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: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.
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

Free camera licenses with purchase of My Cloud NAS

Milestone Arcus software is compatible with thousands of industry-leading cameras for added flexibility. Upon installation on your My Cloud NAS, you will receive two (2) camera licenses already enabled in the software. And for a limited time, get additional camera licenses FREE.

Question has a verified solution.

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

Suggested Solutions

Title # Comments Views Activity
Networking a school 8 85
Intermittent issue reaching a Chinese website. 2 58
Wireshark 7 69
Wi-Fi calling 12 81
Introduction Many times we come across a slowness or instability between two hosts, and almost always we blame the poor networking guys, just because they're an easy target.  Sometimes we forget that other factors including disk bottlenecks, CPU …
Messaging apps are amazing tools with the power to do a lot of good, but the truth is the process of collaborating with coworkers requires relationships established through meaningful communication - the kind of communication that only happens face-…
Internet Business Fax to Email Made Easy - With eFax Corporate (http://www.enterprise.efax.com), you'll receive a dedicated online fax number, which is used the same way as a typical analog fax number. You'll receive secure faxes in your email, fr…
In this tutorial you'll learn about bandwidth monitoring with flows and packet sniffing with our network monitoring solution PRTG Network Monitor (https://www.paessler.com/prtg). If you're interested in additional methods for monitoring bandwidt…

910 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

Need Help in Real-Time?

Connect with top rated Experts

16 Experts available now in Live!

Get 1:1 Help Now