[OpenSIPStack] Sound delay using OSSPhone / Solegy Phone

[Thomas R. <tho...@it...>]

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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