[asterisk-users] Meetme voice quality problems

[Tomasz Zieleniewski]

On Jan 30, 2008 10:35 PM, Dan Austin <Dan_Austin at phoenix.com> wrote:

> Franklin wrote:
> > ztdummy can give you issues as a timing device.
> Yes and no.  See below
>
> > Any way you could try using a Digium card just
> > as a timing device to see if this helps?
>
>
> Tomasz wrote:
> >> I am using Debian OS kernel  2.6.22-3-amd64
> >> and zaptel driver 1.4 with ztdummy module for meetme
> >> application. I use meetme with SIP channels.
>
> Your kernel is new enough that you should be able to
> leverage hi-res timers (you might need to patch ztdummy),
> or at least a RTC set to 8192 ticks/sec.  What does
> dmesg show after ztdummy is loaded?

it is 1024
Zapata Telephony Interface Registered on major 196
Zaptel Version: SVN-branch-1.4-r3748
Zaptel Echo Canceller: MG2
ztdummy: RTC rate is 1024

how can I increase it?

>
>
> >> I have such problem that when one connects to the
> >> conference voice is "cut". Each voice sequence is
> >> disturbed.
> Do you have internal_timing=yes in asterisk.conf?
> This option allows Asterisk to time the RTP stream
> based on zaptel/ztdummy clock and not on the received
> RTP stream.  In a MeetMe, where callers might mute
> themselves, the received RTP stream is all but useless
> for timing.

Yes I have it set.

>
>
> Dan
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080131/45826b7d/attachment.htm