Is conversation half-duplex?

esjones

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Since I've moved one of our POTS lines to PIAF (using a PAP2T-NA) my wife has mentioned that she cannot "cut-in" on another party while they are speaking. Is it normal for VoIP telephony to be one-at-a-time conversation, instead of the ability to speak over the other party in a call?

If this is not normal for VoIP, what can be done to troubleshoot?

Thanks.

Earl
 

rossiv

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I am pretty sure that that is a PAP2 limitation, as opposed to a VoIP limitation. On my Snom M3 speakerphone, when you talk, you can't hear the other person, but the other person hears you. And if the other person talks and you dont, then all is well. That is half-duplex, and also what you are describing.
I think I have a PAP2 or similar, maybe a 3102 or 2102, that I might try to see if I can duplicate what you are experiencing.

Thanks,
 

Linetux

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This is most certainly not a VoIP issue. It's your ATA or endpoint.

But in the same breath I'll tell you that I have not seen this issue with my Linksys ATA's, so perhaps it's a config problem?
 

kmcdaniel

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have to agree on possible config problem. I too do not have this problem with my PAP2t's.
 

esjones

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Thanks for the observations. What config settings might I take a look at?
 

Alex728

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what type of line card and echo canceller are you using to connect your POTS line to PIAF?
 

TheShniz

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I periodically get reports from new installs that Aastra's on Grandstream Gateways also experience half-duplex speech, whereas Grandstream's on Grandstream Gateways don't seem to. Both manufacturers say their respective products are full-duplex, but I have looming in the back of my head that some combinations of product lend to this issue (although I can't see why this would be technically so). If anyone can shed some light, I'd definately be interested to listen.
 

esjones

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Alex. I am not using POTS lines. This is a SIP trunk through Voip.ms.
 

MGD4me

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On my Snom M3 speakerphone...

rossiv...

The 'half-duplex' operation you observed is fairly typical for some speakerphones. When you talk (into the mic) the speaker is muted to avoid feedback, since the mic and speaker are often physically closely positioned. The same 'acoustic coupling' can happen when the other party talks to you, but by adjusting the speaker volume, feedback can be avoided.

The situation for the ATA is different. The 'SPA' series are all full-duplex. I'm wondering (thinking out loud here) if the observed phenomenon is due more to VoIP delay (latency)? That is, when a person 'at this end' (the listener) wishes to break in and talk, that he/she continues to hear the 'far end' talker for a short period of time before his/her voice reaches the far end. Because of this inherent delay, it would 'seem' as though you were not interrupting the talker because their speech continues to be delivered for a half second, or more. This is purely speculation, but it would interesting to determine the amount of latency present between your server and internet telephone service provider.
 

Alex728

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Alex. I am not using POTS lines. This is a SIP trunk through Voip.ms.

In this case its not a PAP2T limitation.

The reason I asked is on PIAF with MG2 echo canceller and some Linksys kit there the comfort noise dropped out on PSTN calls in such a way it felt like it was half-duplex (OSLEC doesn't do this), but I have internal IAX calls between sites (between PAP2Ts and Linksys SPA942) and there is no issue with "half duplex" type calls there..

What does sometimes happen (indeed it happened at my office) even on VOIP calls is when two women with strong voices are talking at the same time, this makes the sound become choppy as something gets overloaded!

I solved this by turning down the FXS port gain slightly...
 

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