Posted: Sun Jul 08, 2007 2:47 pm Post subject: [Asterisk-bsd] Timing issues
Dear list,
We are running FreeBSD 6.2 with Asterisk 1.4.5 (that has been built
with threading in the IAX2 channel removed due to the threading bug in
it). Codec in use before today has been G.729, but we were not able
to get chanspy to work without crashing Asterisk (persistently). So
we switched to G.711, which does not crash asterisk when chanspy is
used. We don't wish to stay with G.711 however, since our VoIP
service provider only support G.729 and thus we want to run G.729 over
the whole network eventually again (all links from the phone, Asterisk
(PBX01), wan link, Asterisk (Vrouter) to VoIP SP).
The biggest problem, however, seems to be a timing issue. This is what happens:
We record all calls in Asterisk, but when the recording of a poor call
(breaking up, bouncing sound, etc) is listened to afterwards, the
quality is good. This, in our minds, points to timing issues, not
codec translation (which is from G.711 to G.729 and finally to GSM
when the call is placed to mobile phones). Strangely enough, the
calls to GSM users sound poor when in progress, but the recordings are
of acceptable quality. Calls to landline numbers though, sound
excellent and recordings are just as good. That seems to indicate
that the timing issues are not just confined to our network, but it's
may be a combination of our and the SP's network and GSM related (if
doesn't make sense though, or does it - that the codec affect the
timing?). Having said that, however, I'm must confess that I'm not
the technical guru that can qualify this suspicion, and I may be wrong
in my analysis of the scenario.
How can we address these issues?
Are they timing issues, or could it be something else?
Are there timing issues in general in Asterisk/FreeBSD if we do not
use any zaptel hardware? (We don't have any legacy PBX or Telco
integration or hardware in our Asterisk installation, only ethernet)
Posted: Sun Jul 08, 2007 3:02 pm Post subject: [Asterisk-bsd] Timing issues
cd /usr/loca/etc/rc.d
and add ztdummy.ko to the end of the module load line and to the begining of
the unload line. this wuill enable ztdummy for timing.
On Sunday 08 July 2007 07:42:17 am Roland Giesler wrote:
Quote:
Dear list,
We are running FreeBSD 6.2 with Asterisk 1.4.5 (that has been built
with threading in the IAX2 channel removed due to the threading bug in
it). Codec in use before today has been G.729, but we were not able
to get chanspy to work without crashing Asterisk (persistently). So
we switched to G.711, which does not crash asterisk when chanspy is
used. We don't wish to stay with G.711 however, since our VoIP
service provider only support G.729 and thus we want to run G.729 over
the whole network eventually again (all links from the phone, Asterisk
(PBX01), wan link, Asterisk (Vrouter) to VoIP SP).
The biggest problem, however, seems to be a timing issue. This is what
happens: We record all calls in Asterisk, but when the recording of a poor
call (breaking up, bouncing sound, etc) is listened to afterwards, the
quality is good. This, in our minds, points to timing issues, not
codec translation (which is from G.711 to G.729 and finally to GSM
when the call is placed to mobile phones). Strangely enough, the
calls to GSM users sound poor when in progress, but the recordings are
of acceptable quality. Calls to landline numbers though, sound
excellent and recordings are just as good. That seems to indicate
that the timing issues are not just confined to our network, but it's
may be a combination of our and the SP's network and GSM related (if
doesn't make sense though, or does it - that the codec affect the
timing?). Having said that, however, I'm must confess that I'm not
the technical guru that can qualify this suspicion, and I may be wrong
in my analysis of the scenario.
How can we address these issues?
Are they timing issues, or could it be something else?
Are there timing issues in general in Asterisk/FreeBSD if we do not
use any zaptel hardware? (We don't have any legacy PBX or Telco
integration or hardware in our Asterisk installation, only ethernet)
regards
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Posted: Sun Jul 08, 2007 7:32 pm Post subject: [Asterisk-bsd] Timing issues
On 2007/07/08 16:42, Roland Giesler wrote:
Quote:
That seems to indicate
that the timing issues are not just confined to our network, but it's
may be a combination of our and the SP's network
aiui, without a timing source outgoing rtp is synced from incoming
rtp, so network problems (jitter, packet loss) on your incoming rtp
will be mirrored in outgoing (silence suppression [VAD] is fun too).
Using zaptel/ztdummy and turning on internal_timing (asterisk.conf)
should help.
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