Re: [OpenSIPStack] Sound delay using OSSPhone / Solegy Phone

[OpenSIPStack F. <ope...@op...>]

Hi,

For the meantime you may uncomment (or put your own range in ms) the line that goes:

SetAudioJitterDelay(100, 1000);

in SoftPhone.cxx.  Its inside the SoftPhoneManager::SoftPhoneManager() constructor block.  

I'll have the setter handy in ATLSIP in a shor while

Joegen


> {quote:title=Guest wrote:}{quote}
> Hi.
> 
> We are currently considering wether we should use OSS for a small client application (possibly using ATLSIP.dll). I've been wondering if it is possible / how difficult it is to change the buffer size because we need smaller delays (this is for LAN  calls only not the internet).
> I've been recording delays of nearly 300ms which is probably ok through the internet, but is a problem for LAN connections at call centers,.. -- I did measure this by enabling recording on our PBX and then using a SIP phone [snom 360] on one end and a PC with Solegy phone / OSSPhone on the other end. At the pc I put a cable from microphone to headset jack (with a condensator and a resistor ;)) so that we can measure the delay properly by just having a look at the wave file.
> 
> Please let me know if this is maybe just a parameter somewhere or not configurable at all.
> 
> Regards,
> 
> Mit freundlichen Grüßen
> Thomas Raschbacher
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