Re: [Ekiga-list] Echo in Calls

[Shawn Adams <shawn_adams web de>]

Robert

I think you've all but proven the issue, if the hard phone works - same
server, same internet connection, same everything - except - the phone
itself (replaced Ekiga with this phone temporarily) and there is no
delay - thus - no echo, the PC is the issue.

Exactly how to track the delay inside the PC from mike/speaker through
ekiga to the network card  - I'll have to defer to the SW guys. I use
lots of older PCs with ancient sound cards, and these usually are a
likely source of latency.

At least you can check the PC->server or PC->PC latency - a simple test
is ping - see what the maximum and average , example, send 5 frames:

sadams tau:~> ping -c 5 ekiga.net
PING ekiga.net (86.64.162.35) 56(84) bytes of data.
64 bytes from ekiga.net (86.64.162.35): icmp_seq=1 ttl=56 time=34.4 ms
64 bytes from ekiga.net (86.64.162.35): icmp_seq=2 ttl=56 time=96.4 ms
64 bytes from ekiga.net (86.64.162.35): icmp_seq=3 ttl=56 time=34.1 ms
64 bytes from ekiga.net (86.64.162.35): icmp_seq=4 ttl=56 time=34.7 ms
64 bytes from ekiga.net (86.64.162.35): icmp_seq=5 ttl=56 time=43.0 ms

--- ekiga.net ping statistics ---
5 packets transmitted, 5 received, 0% packet loss, time 4001ms
rtt min/avg/max/mdev = 34.106/48.542/96.460/24.190 ms

check to see if the PC network card has errors:
netstat -s
ifconfig
ethtool -S eth0 /mii-diag (depends on card/distro)


Note the max and average delay. anything over approx. 25ms - You have a
chance of hearing the echo. Now how to do this between application and
network interface - I'll have to pass.

Hope this helps.




Robert Ingraham wrote:
>
> After fiddling around a little bit, I think the real problem is not so
> much echo as latency, and I think the problem is probably with the
> Ekiga/Jaunty combination.  I set the system audio settings and the
> Ekiga audio settings both to the same thing, in this case to
> Nvidia/Alsa (my sound card is Nvidia), and I lowered the jitter buffer
> from 500 (the default) to 220. This eliminated a lot (not all) of the
> echo.  If I lower the jitter buffer any more, the calls begin to break
> up.  However, there is still a very bad latency (delay) problem. Very bad.
>
> As a test, I unplugged my laptop from the ethernet wire, and plugged
> the same ethernet wire into a VOIP hard phone, configured to use the
> same server.  On the VOIP hard phone, there is no echo, no delay, and
> the calls are clear.  No noticible latency problem at all.
>
> I should mention that I used Ekiga under Ubuntu Intrepid, and I did
> not have the Latency/Echo problems.  It only started after I upgraded
> to Jaunty beta.  Any thoughts?
>
>
>
> On Sat, Apr 4, 2009 at 3:06 AM, Shawn Adams <shawn_adams web de
> <mailto:shawn_adams web de>> wrote:
>
>     Robert,
>
>     It's not a bandwidth issue.
>
>     There are two main sources of echo in telephony networks: acoustic
>     echo
>     and line echo. Acoustic echo is generated on any phone (IP phone or
>     analog) when there is feedback from the speaker to the microphone.
>
>     This is particularly noticeable on many speaker phones. Line echo is
>     very common in the PSTN network, and this most commonly occurs when
>     there is a two-wire to four-wire conversion in the network (for
>     example,
>     where analog is converted into T1 or E1). In the good old telephone
>     network, the delay is so small, the human ear doesn't notice.
>
>     They have also 50+ years experience with echo cancellation, which
>     works
>     well on analog signals, but is more difficult with VOIP.
>
>     Leak-through happens only in analog circuits. VOIP frames do not leak.
>
>     Echo is almost always happening at the opposite end of where the
>     echo is
>     perceived. It might be happening at the PSTN gateway as well.
>
>     If you've set your mute button, but the other side still hears
>     echo, it
>     is possibly a leak in your VOIP phone or sound card/drivers area,
>     or the
>     gateway.
>
>     The gateway to the PSTN is just another source of delay, they more
>     delay
>     - the more perceptable the echo.
>
>
>     #1 does a purely SIP/VOIP call have the echo  ?
>
>     - if NO - the echo may be induced at the PSTN gateway, in each case,
>     check the round trip delay between your phone and the target.
>
>     #2 what is the round trip delay from your phone to the PSTN gateway ?
>
>     - there is almost always echo on a good old fashioned telephone,
>     but you
>     don't notice it because the delay is <25ms
>
>     - if you can somehow reduce the round trip delay - this might minimize
>     the perceived echo, 150ms is usually a good rule-of-thumb as maximum
>     tolerable for VOIP calls.
>
>     #3 it is possible the echo is in the sound card/devices/drivers - I've
>     had luck playing with volume levels disabling of the various sound
>     card
>     devices - capture device, PCM volume, etc...
>
>     So see if you can isolate whether the echo is at the gateway, or
>     at your
>     phone. There may be little you can do about the gateway, except to
>     complain to the provider. If it's at the phone - take a look at the
>     various devices/pulse audio, etc...
>
>     My 2 cents....
>
>
>     Hope this helps.
>
>     Best regards,
>
>     Robert Ingraham wrote:
>     > I have read previous posts on this subject, but I still cannot
>     resolve
>     > the problem. I use Ekiga to make PC-to-Phone calls. I use a headset.
>     > I am running Ubuntu Jaunty beta, with PulseAudio. When I make
>     calls, I
>     > can hear fine, with no echo, but the people I am talking to
>     complain of
>     > a very bad echo. This includes both landlines and cellphones.
>     There is
>     > also a delay between when they talk, and when I hear them. The
>     headset
>     > I use has a mute button, but even if I mute my end, the other person
>     > still hears an echo. I have tried enabling both "echo
>     cancelation" and
>     > "silence detection" in Ekiga's preferences, but this has had no
>     effect
>     > at all on the echo. Ekiga's codec preferences are set to use Speex
>     > 16khz codec. I have a very good Comcast cable connection, with
>     download
>     > rates in the 750mb-1gb range. Anyone have any idea how to fix this?
>     > Would turning off PulseAudio help?
>     >
>     >
>     >
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-- 
Shawn Adams
shawn_adams web de