Audio bypass: audio directly between phone & provider

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ilkeryildiz

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I see below information other ip-pbx web site.

How does 3cx work for transmits audio between external(remote) extension-phone & Voip-provider?
How does 3cx work for transmits audio between external(remote) extension-phone & local extensions?
directly or via 3cx?

Audio bypass:
"This is an advanced option. When setting to yes be sure to make some testcalls. Audio bypass transmits audio directly between phone & provider if both are enabled. Some devices and providers don't support it resulting in dropped calls or missing audio."

do you add this option?
thanks.
 
Hi,

VoIP provider calls the 3CX as a RTP Proxy.

Internally the PBX does not act as a proxy, though extensions, and even gateways CAN be configured to 'bind to MS', i.e. use the PBX for RTP.
 
I tested bandwidth use with "Bandwidth Meter Pro" program.

my pc use 80-85 kbps bandwidth when call between external(remote) extension-phone & local extension.

my pc use 170-180 kbps bandwidth when call between external(remote) extension-phone & Voip-provider.

my pc use 80-85 kbps bandwidth when call between internal extensions and Voip-provider

all audio transmits via my computer if extension outside internal (remote location).

If I checked "external device" option on add extension page,
are transmits audio directly between external extension-phone & Voip provider?

I want transmits audio directly between external extension-phone & Voip provider when extension outside internal.
is this possible?

sorry my english very bad
 
Hi,

If you will not bound both external extension and VoIP provider to Media Server (see corresponding settings) -- they will try direct RTP connection. But they both should support re-INVITE, otherwise they will be explicitly bound to Media Server. We can not guarantee that you will hear audio in that case, external extension should be able to resolve and provide in SDP externally accessible RTP address:port.
 
archie said:
Hi,

If you will not bound both external extension and VoIP provider to Media Server (see corresponding settings) -- they will try direct RTP connection. But they both should support re-INVITE, otherwise they will be explicitly bound to Media Server. We can not guarantee that you will hear audio in that case, external extension should be able to resolve and provide in SDP externally accessible RTP address:port.

How can i set "corresponding settings" and " not bound both external extension and VoIP provider to Media Server" ?
where is it this setting at 3cx?

"Edit extension > Advanced options > Bind to Media Server" check box
or
Edit VOIP Provider > Advanced options > Bind to Media Server" check box

I tried both check box uncheck and my computer(3cx) not use bandwith when call between external(remote) extension-phone & Voip-provider
very good

3cx log:

18:50:59.921 StratLink::onHangUp Call(C:F): got Hang-Up from Ext.101; reason: BYE
18:49:09.671 CallLegImpl::onConnected Established media channel for Ln:10002@borsamatik@lowratevoip: remote=194.120.0.163:25432; local=81.213.108.235:5100
18:49:09.671 StratInOut::onConnected Call from Ext.101 to Ln:10002@borsamatik@lowratevoip is established
18:49:09.671 CallLegImpl::onConnected Established media channel for Ext.101: remote=81.213.108.235:5100; local=194.120.0.163:25432
18:49:01.437 CallConf::onIncoming Incoming call from Ext.101 to sip:[email protected]
 
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