A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
having a conversation. Call quality is reported as good except for an
echo with a 3 second delay.
Most of my searches are saying echo happens only on the PSTN piece, but
there isn't one here.
Can someone point me in the right
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
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.
Also, do a ping to see latency, 200ms is pretty much my standard.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
On Thu, Nov 20, 2008 at 12:16
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
Steve Totaro wrote:
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.
Also, do a ping to see latency, 200ms is pretty much my standard.
Coming from outside the network, setting up for a couple rounds of
NATting isn't going to work
On Thu, Nov 20, 2008 at 1:13 PM, c james [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.
Also, do a ping to see latency, 200ms is pretty much my standard.
Coming from outside the
There are also settings which will turn on local echo cancellation for
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
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.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
On
c james 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 an
echo with a 3 second delay.
Most of my searches are saying echo happens only on the PSTN piece, but
there isn't one here.
Which end
Ok, I'll bite, what possible IAX bugs/shortcomings/features can cause
echo ?
Tim.
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
c james [EMAIL PROTECTED] writes:
A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
having a conversation. Call quality is reported as good except for an
echo with a 3 second delay.
Feedback from speaker to microphone. The problem is always at the end
which doesn't hear it.
Drew Gibson wrote:
c james 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 an
echo with a 3 second delay.
Most of my searches are saying echo happens only on the PSTN piece, but
there isn't one
Steve Totaro wrote:
On Thu, Nov 20, 2008 at 1:13 PM, c james [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.
Also, do a ping to see latency, 200ms is pretty much my standard.
Coming
Coming from outside the network, setting up for a couple rounds of
NATting isn't going to work well. They are not seeing it between
phones. Others, using the polycom phones have reported echo between two
SIP on a 4ms ping trip.
Could this be due to a purely acoustic echo within the Polycom
On Thu, Nov 20, 2008 at 6:27 PM, Dave Platt [EMAIL PROTECTED] wrote:
Coming from outside the network, setting up for a couple rounds of
NATting isn't going to work well. They are not seeing it between
phones. Others, using the polycom phones have reported echo between two
SIP on a 4ms ping
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