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[asterisk-users] lots of warnings from translate.c


 
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francesco.castellano a...
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PostPosted: Tue Apr 22, 2008 5:28 am    Post subject: [asterisk-users] lots of warnings from translate.c Reply with quote

We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10,
acting as gateways from SIP to ISDN PRI interfaces. Each has one
Digium TE420 (with hardware echo cancellation) and one TC400B for
transcoding, in order to handle 60/90 contemporary calls in peak
hours.

In my logs there are hundreds of thousand warnigs per day like these:

transcode.c: no samples for lintoulaw
transcode.c: zapg729toalaw did not update samples ###

Could you suggest me what are the possible causes for that? Are they
signs of bad audio quality? Any ideas for resolving these issues?

In addition I can say that we are using a quite large jitter buffer in
zapata.conf:

jitterbuffers=16 (=> 0.32s)

Moreover, it uses the fixed implementation, because when I tried the
adaptive one I experienced one-way audio.
Finally I have to note that, using a Siemens IP phone (G.729 no
AnnexB) in conditions of no load on servers, I could replicate
non-deterministically (sigh!) each of these problems, with a very
noisy audio, and a annoying period of silence during the first seconds
of call.

Regards,
Francesco

PS. Previous versions of asterisk and zaptel presented an identical situation.
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dboyd at ignitetrx.com
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PostPosted: Tue Apr 22, 2008 5:50 am    Post subject: [asterisk-users] lots of warnings from translate.c Reply with quote

On Tue, 2008-04-22 at 12:28 +0200, Francesco Castellano wrote:
Quote:
We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10,
acting as gateways from SIP to ISDN PRI interfaces. Each has one
Digium TE420 (with hardware echo cancellation) and one TC400B for
transcoding, in order to handle 60/90 contemporary calls in peak
hours.

In my logs there are hundreds of thousand warnigs per day like these:

transcode.c: no samples for lintoulaw
transcode.c: zapg729toalaw did not update samples ###

Could you suggest me what are the possible causes for that? Are they
signs of bad audio quality? Any ideas for resolving these issues?

In addition I can say that we are using a quite large jitter buffer in
zapata.conf:

jitterbuffers=16 (=> 0.32s)

Moreover, it uses the fixed implementation, because when I tried the
adaptive one I experienced one-way audio.
Finally I have to note that, using a Siemens IP phone (G.729 no
AnnexB) in conditions of no load on servers, I could replicate
non-deterministically (sigh!) each of these problems, with a very
noisy audio, and a annoying period of silence during the first seconds
of call.

Regards,
Francesco

PS. Previous versions of asterisk and zaptel presented an identical situation.

Have you tried additional types of phones and if so can you produce the
same non-deterministic problems?
Dave
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francesco.castellano a...
Guest





PostPosted: Tue Apr 22, 2008 9:37 am    Post subject: [asterisk-users] lots of warnings from translate.c Reply with quote

I tried calls with a couple of other IP phones, but not extensively,
and these warnings did not happen. Anyway, also with the Siemens IP
phone, the warnings happen only sometimes.

At the moment, I'm not succeeded in correlating the warnings to a
specific user-agent.

Thanks,
Francesco

On Tue, Apr 22, 2008 at 12:50 PM, David Boyd <dboyd at ignitetrx.com> wrote:
Quote:

On Tue, 2008-04-22 at 12:28 +0200, Francesco Castellano wrote:
Quote:
We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10,
acting as gateways from SIP to ISDN PRI interfaces. Each has one
Digium TE420 (with hardware echo cancellation) and one TC400B for
transcoding, in order to handle 60/90 contemporary calls in peak
hours.

In my logs there are hundreds of thousand warnigs per day like these:

transcode.c: no samples for lintoulaw
transcode.c: zapg729toalaw did not update samples ###

Could you suggest me what are the possible causes for that? Are they
signs of bad audio quality? Any ideas for resolving these issues?

In addition I can say that we are using a quite large jitter buffer in
zapata.conf:

jitterbuffers=16 (=> 0.32s)

Moreover, it uses the fixed implementation, because when I tried the
adaptive one I experienced one-way audio.
Finally I have to note that, using a Siemens IP phone (G.729 no
AnnexB) in conditions of no load on servers, I could replicate
non-deterministically (sigh!) each of these problems, with a very
noisy audio, and a annoying period of silence during the first seconds
of call.

Regards,
Francesco

PS. Previous versions of asterisk and zaptel presented an identical situation.

Have you tried additional types of phones and if so can you produce the
same non-deterministic problems?


Dave


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