[asterisk-users] SIP to IAX2 with delayed echo
thp at westhawk.co.uk
Thu Nov 20 15:00:22 CST 2008
Ok, I'll bite, what possible IAX bugs/shortcomings/features can cause
On 20 Nov 2008, at 18:47, Steve Totaro wrote:
> Simple tests. Change from the highly touted "IAX2" to SIP, but before
> that, download X-Lite and see if you have the same delay. If you
> don't then look at your Polycoms, if you do, then switch to SIP.
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
> On Thu, Nov 20, 2008 at 1:39 PM, Dave Fullerton
> <dfullertasterisk at shorelinecontainer.com> wrote:
>> There are also settings which will turn on local echo cancellation
>> the handset, headset and/or speaker phone. I don't recall their
>> names at
>> the moment. They are off by default on the handset and headset unless
>> you're using a very recent (3.0+) SIP app.
>> Tim Nelson wrote:
>>> I'm not sure about the 3 second delay, but I've seen plenty of
>>> echo issues on Polycom phones when the gain has been changed on
>>> the handset. Check the voice.gain.tx and voice.gain.rx settings in
>>> your sip.cfg to make sure they're not too high.
>>> You also may want to make sure there aren't any system resource
>>> constraints such as high CPU usage or memory usage... :-)
>>> Tim Nelson
>>> Systems/Network Support
>>> Rockbochs Inc.
>>> (218)727-4332 x105
>>> ----- "c james" <cjames at callone.net> wrote:
>>>> A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
>>>> having a conversation. Call quality is reported as good except for
>>>> echo with a 3 second delay.
>>>> Most of my searches are saying echo happens only on the PSTN piece,
>>>> there isn't one here.
>>>> Can someone point me in the right direction?
>>>> Asterisk 220.127.116.11
>>>> Under 40 users
>>>> Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM (Hey, that's
>>>> they wanted to use!)
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