Problem with MRCPSynth on Hyper-V

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Mulai

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May 30, 2012, 11:40:41 AM5/30/12
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Hello Arsen,

I try to use Asterisk+UniMRCP on Hyper-V Guest OS (CentOS 5 and 6). I
installed Integration Tools and everything is working correct.
But I have problem with synthesis.
When I'm using MRCPSynth in dialplan, can be heard noises, cracks and
gnashing.
If I wrote sound on MRCP-server, it's OK. Problem is in Asterisk.
I didn't see errors in logs or Asterisk's console.

May you tell what could be the problem?

Arsen Chaloyan

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Jun 6, 2012, 8:38:30 PM6/6/12
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Hello Kirill,

First, I'd check RTP statistics logged at the end of each MRCP session
as follows

Close RTP Receiver [r:110 l:0 j:0 p:50 d:0 i:0],

where
r: received
l: lost
j: jitter
p: playout delay
d: dropped
i: ignored

Next, if you're using r1835 or later, then I'd suggest to try the
adaptive mode with the following parameters, for example,

<jitter-buffer>
<adaptive>1</adaptive>
<playout-delay>50</playout-delay>
<max-playout-delay>1000</max-playout-delay>
<time-skew-detection>1</time-skew-detection>
</jitter-buffer>
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Arsen Chaloyan
Author of UniMRCP
http://www.unimrcp.org

Mulai

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Jun 7, 2012, 6:35:22 AM6/7/12
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Arsen, thanks.

I use asterisk-unimrcp-0.3.2.
May you explain how can I check RTP statistics? I don't know what is
RTP Receiver and how I can use it.

Where is jitter-buffer parameters you are writing about?
I'm using mrcp.conf for all settings.
Default is:
; --- Jitter buffer settings ---
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200

There are cracks always when synthesis is starting.
If I increase playout-delay cracks become louder and stronger. playout-
delay = 10 cracks are softer but still heard.
> > For more options, visit this group athttp://groups.google.com/group/unimrcp?hl=en.
Message has been deleted

Arsen Chaloyan

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Jun 15, 2012, 5:54:02 PM6/15/12
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Hi Kirill,

On Thu, Jun 7, 2012 at 3:35 AM, Mulai <kir....@gmail.com> wrote:
> Arsen, thanks.
>
> I use asterisk-unimrcp-0.3.2.
> May you explain how can I check RTP statistics? I don't know what is
> RTP Receiver and how I can use it.

You can check the mentioned RTP stats from the log file.

Set log-level in /etc/asterisk/mrcp.conf. to debug

; UniMRCP logging level to appear in Asterisk logs. Options are:
; EMERGENCY|ALERT|CRITICAL|ERROR|WARNING|NOTICE|INFO|DEBUG -->
log-level = DEBUG


If you want them to see on console of Asterisk, set the following in
/etc/asterisk/logger.conf

console => notice,warning,error,debug


>
> Where is jitter-buffer parameters you are writing about?
> I'm using mrcp.conf for all settings.
> Default is:
> ; --- Jitter buffer settings ---
> playout-delay = 50
> ; min-playout-delay = 20
> max-playout-delay = 200

They are not available with the version you're using.

>
> There are cracks always when synthesis is starting.
> If I increase playout-delay cracks become louder and stronger. playout-
> delay = 10 cracks are softer but still heard.

I need to see the incoming RTP stream to make any comments. Wireshark
capture might be helpful.
> For more options, visit this group at http://groups.google.com/group/unimrcp?hl=en.

Mulai

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Jun 19, 2012, 9:38:03 AM6/19/12
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Hi Arsen,

In attach is log Asterisk from console with enabled rtp debug and debug settings in logger.conf.
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>> Arsen Chaloyan
>> Author of UniMRCPhttp://www.unimrcp.org
>
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broken_tts.txt

Mulai

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Jun 19, 2012, 10:32:37 AM6/19/12
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In attach dump by tcpdump.


On Saturday, June 16, 2012 1:54:02 AM UTC+4, Arsen Chaloyan wrote:
>> > To unsubscribe from this group, send email to unimrcp+unsubscribe@googlegroups.com.
>> > For more options, visit this group athttp://groups.google.com/group/unimrcp?hl=en.
>>
>> --
>> Arsen Chaloyan
>> Author of UniMRCPhttp://www.unimrcp.org
>
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broken_tts.dmp

Arsen Chaloyan

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Jul 27, 2012, 7:23:08 PM7/27/12
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Hi Kirill,

I'd like to bring your attention to the media path and codecs.

[Jun 19 23:22:35] DEBUG[2589]: app_unimrcp.c:4172 unimrcp_log: Media
Path 0x2dbf8b8 Source->[PCMU/8000/1]->Decoder->[LPCM/8000/1]->Bridge->[LPCM/8000/1]->Encoder->[PCMA/8000/1]->Sink

It looks like PCMU was used on the MRCP leg while the inbound call was
established with PCMA. This shouldn't cause any problem in theory, but
try to avoid the conversion by using the same codec on both legs. The
cracks you heard may come from the conversion, I guess. Otherwise, the
RTP stream wasn't that bad.
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>> >> Arsen Chaloyan
>> >> Author of UniMRCPhttp://www.unimrcp.org
>> >
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>>
>>
>> --
>> Arsen Chaloyan
>> Author of UniMRCP
>> http://www.unimrcp.org
>
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Mulai

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Jul 28, 2012, 3:20:30 AM7/28/12
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Thanks Arsen.

>> >> > For more options, visit this group
>> >> > athttp://groups.google.com/group/unimrcp?hl=en.
>> >>
>> >> --
>> >> Arsen Chaloyan
>> >> Author of UniMRCPhttp://www.unimrcp.org
>> >
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>> >
>>
>>
>>
>> --
>> Arsen Chaloyan
>> Author of UniMRCP
>> http://www.unimrcp.org
>
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