Voice quality problem on the simultaneous second call

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Jim Lam

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Hi Everyone,

I got a weird voice quality problem as mentioned above:

Situation:
- When a call is made/received - if it is the first call on the site - everyone works perfectly - great voice quality
- If at the same time, someone make another call from another extension - the voice quality absolutely dropped to a point of unacceptable - cackling noise, only hear some part of the sentence - BUT it only happens to the second call (the first call continues to be of great quality). And also the voice quality issue ONLY happen on the remote end, the local end is 99% fine.

- Bandwidth is not an issue, we are on a pretty fast connection.
- PBX server resource is not an issue, CPU/Network utilisation is typically under 10%
- Ping time between site and PBX is less than 10ms, and ping time between PBX with upstring VoIP provider is less than 20ms

I think I may have a problem with the router setting or phone setting, but I am not sure how to troubleshoot.

Thanks for your help in advance.
Jim
 
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Jim Lam

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Apr 5, 2017
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The local end is on ALAW, the PBX is on ALAW as well as the VoIP provider.

We have tried the remote end either using both PSTN line and mobile phone.

It has the same result. The first call went thru ok, the second simultaneous call got crackling and lost packets.
 
EasyBB

EasyBB

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Oct 23, 2016
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Jim Lam said:
someone make another call from another extension - the voice quality absolutely dropped to a point of unacceptable
Click to expand...
Jim Lam said:
Bandwidth is not an issue, we are on a pretty fast connection.
Click to expand...
Jim Lam said:
The local end is on ALAW, the PBX is on ALAW as well as the VoIP provider.
Click to expand...

I probably can help; but before getting into the details could you please state the country, ISP, type of internet and voice provider?
 
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Jim Lam

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Apr 5, 2017
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Thanks for your offer.

To answer your question, we are in Australia, using Telstra with a NBN connection with speed of 100Mbit down and 40Mbit up. We are connecting to our own Fusion PBX and our VoIP providers are Faktortel and Siptalk.

Regards
 
EasyBB

EasyBB

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Oct 23, 2016
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Jim Lam said:
we are in Australia, using Telstra with a NBN connection with speed of 100Mbit down and 40Mbit up
Click to expand...
Yes, I thought so :)
Your connection through Telstra is on a DSCP profile which has only 150kbps for VoIP. Assuming one ALAW call uses around 85kbps, the second call would only work if you get at least 170kbps TC-1 traffic class. You have many options to solve the issue. Which router do you use?
 
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Jim Lam

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Apr 5, 2017
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You are serious??? I think I have set the DSCP to be on best effort instead of expedited.

Anyway, I am using a Draytek Vigor 2860ac.
 
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Jim Lam

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Apr 5, 2017
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Just tried the following:

The script return all passed, so that seems to be ok. I have tried both from an internal box to my Fusion PBX in the cloud, and from my cloud PBX to faktortel's asterisk box, all replied passed.

To be doubly certain, I have go into all of my Yealink phones and set both Voice QoS and SIP QoS to 0.

Unfortunately the problem still persists.:(:(:(

Any more ideas??
 
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Jim Lam

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Apr 5, 2017
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Hey EasyBB

Just have a more detailed look at your script and the message, it only tests DSCP based routing, NOT bandwidth allocation.

Assuming you are correct that Telstra is rate limiting on the DSCP value, the ping will work fine. Problem only arise when the bandwidth utilisation exceeds the allocated value.
 
EasyBB

EasyBB

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Oct 23, 2016
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Jim Lam said:
Just have a more detailed look at your script and the message, it only tests DSCP based routing, NOT bandwidth allocation.
Click to expand...
Correct; there can be two issues with Telstra NBN connections- the first one where Telstra gobbles up all non DSCP0 packets and the second where TC-1 bandwidth is limited to 150kbps and your connection profile has DSCP mapping. My script only checks the first one while the second is evident from the problem you are having.
 
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Jim Lam

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Apr 5, 2017
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That's really bad. Do you know whether other NBN reseller have such restrictions? For example, Optus/TPG/iiNet or whoever.

And then it seems for the moment I have to try to use a codec with a lower bandwidth requirement.
 
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Jim Lam

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Apr 5, 2017
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Oh, another question. If I have set all the packet to DSCP 0, Telstra should not be able to differentiate between them, and in that case my VoIP traffic should be fine??

So in that case, it probably means that I have NOT set all packets to DSCP 0. But that should not be right, as I have already set the QoS value to 0 in the Yealink phone itself.

Any ideas??
 
EasyBB

EasyBB

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Oct 23, 2016
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Australia
Jim Lam said:
And then it seems for the moment I have to try to use a codec with a lower bandwidth requirement.
Click to expand...
You have a few options, actually more than a few
  1. Change provider. Try aussiebroadband, Telecube (siptalk) etc
  2. Configure Draytek to remark all packets to DSCP 0
  3. Get another router which is easier to configure
  4. Use g729 codec
  5. Ask Telstra to move you from DSCP mapped profile to Default profile
 
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Jim Lam

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Apr 5, 2017
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The Telstra mapped profile option and the G729 codec option seems to be most attractive.

Does Telstra entertain such requests normally?
 
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