[asterisk-users] SPA941 -> Asterisk -> Voip provider -> PSTN ->
cliff.brake at gmail.com
Wed Sep 27 12:36:44 MST 2006
On 9/22/06, Rich Adamson <radamson at routers.com> wrote:
> > So, it seems there is some type of weird interaction between my system
> > and the ShoreTel system if I use the SPA941 IP phone.
> > Does anyone have suggestions as to how I can start debugging this?
> Check the RTP Packet Size (under the Sip tab). Set it to .020 (20
> milliseconds) and place another test call. For whatever reason, the
> Linksys/Sipura products default to 30 milliseconds and has impacted the
> quality of audio on some systems.
Setting the RTP packet size to 20ms seems to have fixed it. Thanks
for the suggestion.
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