Call Audio Buffer on outbound - 1 or 2 secs

[loowee]

Hey guys !

We have a system that's been running quite good for 2 months now but the client is reporting a strange thing we cant fix.

the outbound calls have a latency when the remote callee picks up the call and says "hello", we are missing the first words being said for 1 or 2 seconds, this is very annoying cause we miss the greetings of the callee. the strangest part of this problem is that it does not do it with an older generation of Cisco phones( spa942 ) or the 3CX Windows phone.

the phones are CISCO spa303G and the jitter or packets settings are as recommended by 3CX, we also tried all combinations of jitter and packets but we can only get the buffer or lag worst. We also tried PBX delivers audio and other recommendations from the forum with no success.

We are using a VOIP provider service for outbound lines ( broadconnect - Canada ) and a PSTN Gateway for inbound ( grandstream GXW-4104 ). 3CX is on version 10 and configured like all our other installs.

Is there a way to reduce the buffer on the outbound calls ?
I noticed there is record buffer and play buffer on the 3CX phone app, is there an equivalent on hardware phone such as the SPA303g ?

Note that inbound calls also suffer from a minor buffer lag when local receptionnist answers but it is much less than the outbound one. This might be due to the fact we use different services for inbound and outbound.

anyone ?
Thx !

 

[mixig]

Hi there,

we had exactly the same problem as you.. what we found on some forum is that is this:

trunk:
'supports re-invite' - off
'Supports replaces header' off
'PBX delivers audio' - on

extension:
'supports re-invite' - on
'Supports replaces header' off
'PBX delivers audio' - off

trunk settings are default, changes are on extension settings... after that there was no initial rtp delay anymore....

Also check you CPU. Is 3cx maybe on virtual machine?

 

[loowee]

I will try these settings tomorrow morning, I want to be on the spot to make sure it does not affect reliability on the system.
The only thing that is to change on our end to get these setting is the 'Supports replaces header' off that is on for all extensions.

Hope this does it, been trying many things but no real difference.

Thank you !

 

[loowee]

We found the culprit.

It seems that the switches cascade the client has adds a lag in the network.
Net transfert speeds are fine and we did not notice any other latency than with the initial buffering of the outbound calls. What's weird is that SIP does not require 100MBs to be stable, isnt there some way to accelerate the performance trough the switches without having to change wiring structure ?

We will be there tomorrow and see what we can do for the cable structure but this one was not built by our team so it would be easier to change compression, jitter or buffer settings on the software and telephone side if any possible.

Any idea ?

 

[mixig]

if you have managed switches you can prioritize traffic, add phones to Voice VLAN