PBX Delivers Audio Help

[LCC]

Can someone please give me a quick lesson on this? I get that if I check it, then the PBX is processing the audio. Is this a good thing or a bad thing? I want to use G.722. My phones support it. I have v10 sp1 on a beast of a machine. I have a Gigabit network. I'm just not sure what is the best way to implement. My VOIP provider (callcentric) does not support G.722, but that doesn't stop me from using it internally and I may be switching. PLease help a newbie out.

Thanks.

 

[leejor]

As long as 3CX supports the codec, it will translate between devices (a provider) that does not support what you have forced your local device to use. In an ideal situation, to reduce traffic through 3CX, once the call was set-up, all voice traffic would just flow between the individual SIP devices and not have to pass through 3CX.

 

[LCC]

As long as 3CX supports the codec, it will translate between devices (a provider) that does not support what you have forced your local device to use. In an ideal situation, to reduce traffic through 3CX, once the call was set-up, all voice traffic would just flow between the individual SIP devices and not have to pass through 3CX.

Thanks. Starting to make more sense. I just looked at the setting on the VOIP provider and PBX delivers audio is checked, but not checked on my individual extensions. This makes sense now.

*edit*
I stand corrected, it looks like CallCentric does support G.722...

 

[willow]

PBX delivers audio is godd for remote phones that if they tried to send the audio directly would not find the other end and thus need the pbx to act as a middle man. Ideally, you would like the endpoints on a local network to send the audio directly to each other.

 

[leejor]

PBX delivers audio is godd for remote phones that if they tried to send the audio directly would not find the other end and thus need the pbx to act as a middle man.

Although, it can depend on how and where remote extensions are set up. Two at the same location, behind the same router, or perhaps two sets, at different locations but in one country while the 3CX server is in another. In some cases you can reduce latency (dramatically) by not having the audio go all the way back to 3CX, and return.

Depending on the particular set-up (with remote extensions), it may work just fine, or, you can have issues. It is best if you can get it working so that audio passes directly between devices in as many cases as possible/practical. In an ideal situation, 3CX will set-up the call and then drop out leaving the two devices to "talk amongst themselves".