Open Source Sip: [OpenSIPStack] Cpmfort Noise Support ...

9 Replies Last post: Jun 5, 2007 1:03 AM by Guest

Guest

May 29, 2007 5:29 AM

[OpenSIPStack] Cpmfort Noise Support

Hi All,First, Thanks a lot for all the great job you are doing for OpenSipStack and AtlSIPI''m doing some devlopement test with the Softphone ActiveX, the quality is very good and no bugs detected, the only thing is that the softphone is doing by default some VAD and it is not transmiting the silence, so there is no Comfort Noise generation sent whene the calling party stop talking. i heard about a new ActiveX version which will be available and gives the option to enable or disable CNG, is it ready? if yes can i have it please?Other Thing, on my Asterisk Server only G729 Work and not G729AWhat''s Wrong ?
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Guest

1. May 29, 2007 6:49 AM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Hi,

We haven''t exposed this yet but we will soon. Please wait for updates in
this list.

For the meantime, please refer to the attached email on how this can be
done.

Thanks.

Regards,
Ilian

Yacine Auczone wrote:

Hi All,
First, Thanks a lot for all the great job you are doing for
OpenSipStack and AtlSIP
I''m doing some devlopement test with the Softphone ActiveX, the
quality is very good and no bugs detected, the only thing is that the
softphone is doing by default some VAD and it is not transmiting the
silence, so there is no Comfort Noise generation sent whene the
calling party stop talking. i heard about a new ActiveX version which
will be available and gives the option to enable or disable CNG, is it
ready? if yes can i have it please?
Other Thing, on my Asterisk Server only G729 Work and not G729A
What''s Wrong ?

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Guest

2. May 29, 2007 7:52 AM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Hi Ilian,

Can you provide an ETC for exposing Jitter and Silent Detection params
in ATLSIP? Seems like a popular request.

Joegen

Ilian Jeri C. Pinzon wrote:

Hi,

We haven''t exposed this yet but we will soon. Please wait for updates
in this list.

For the meantime, please refer to the attached email on how this can
be done.

Thanks.

Regards,
Ilian

Yacine Auczone wrote:

Hi All,
First, Thanks a lot for all the great job you are doing for
OpenSipStack and AtlSIP
I''m doing some devlopement test with the Softphone ActiveX, the
quality is very good and no bugs detected, the only thing is that the
softphone is doing by default some VAD and it is not transmiting the
silence, so there is no Comfort Noise generation sent whene the
calling party stop talking. i heard about a new ActiveX version which
will be available and gives the option to enable or disable CNG, is it
ready? if yes can i have it please?
Other Thing, on my Asterisk Server only G729 Work and not G729A
What''s Wrong ?

_______________________________________________
opensipstack-devel mailing list
opensipstack-devel at lists dot sourceforge dot net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

No virus found in this incoming message.
Checked by AVG Free Edition. Version: 7.5.472 / Virus Database:
269.8.1/822 - Release Date: 5/28/2007 11:40 AM

Subject:
[Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise]
From:
"Joegen E. Baclor"
Date:
Tue, 29 May 2007 18:26:21 +0800
To:
"Ilian Jeri C. Pinzon"

To:
"Ilian Jeri C. Pinzon"

Subject:
Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise
From:
"Joegen E. Baclor"
Date:
Thu, 12 Apr 2007 16:40:04 +0800
To:
opensipstack-devel at lists dot sourceforge dot net

To:
opensipstack-devel at lists dot sourceforge dot net

Whit,

Good to hear you nailed it! Can''t wait to see your contributions if
you get the chance to expose the other setters/accessors in ATLSIP.

Whit Thiele wrote:

Joegen,

Thanks for the help. I thought I''d send the list an update on what
solved my
problem. I changed the Silence Detector to Fixed with a threshold of
3. This
eliminated all the problems! It seems that the adaptive silence
detector was
constantly incrementing and started affecting things about 10-15 seconds
into a conversation!

I''ll probably put in the ability to change the jitterbuffer and silence
detector into the ATLSIP library and send this in to the project in
the next
couple weeks...

Whit

-----Original Message-----
From: opensipstack-devel-bounces at lists dot sourceforge dot net
[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On Behalf Of
Joegen E. Baclor
Sent: Tuesday, April 10, 2007 9:36 PM
To: opensipstack-devel at lists dot sourceforge dot net
Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise

Whit,

It is probably best to ask this question to
openh323-devel at lists dot sourceforge dot net. However, here''s how to set
the silence detection in code.

OpalSilenceDetector::Param param;
param.Mode = OpalSilenceDetector::NoSilenceDetection;
sfManager.SetSilenceDetectParams( params );

Hope that helps.

dev at whit dot ca wrote:

Joegen,

Thanks for the reply. I''ve been trying different jitterbuffer
settings as

well

as changing the number soundChannelBuffers to a number of different

settings

which I came across in some online
Opal documentation (

http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html

)

I''ve tried setting the jitter buffer to minimums 25 through to 500
and the

depth

to as high as 15 but nothing is helping. As I described before, I
can get

about

10-15 consecutive seconds of decent voice quality and then it gets very

choppy.

Is anyone else experiencing this?

I am wondering if it may have something to do with the Silence
detection

portion

of Opal. I''ve noticed in the opal.log file that the Silence Threshold

creeps

upwards the longer the person talks. Is there a way to disable the
silence
detector? I could see that there are several Modes (Fixed, Adaptive,
etc)

for

it but I can''t figure out where this is initialized in the code.
I may be on the wrong track but I can''t figure out this strange
behavior.
Any help/ideas/suggestions would be greatly appreciated!

Whit

-----Original Message-----
From: opensipstack-devel-bounces at lists dot sourceforge dot net
[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On Behalf Of

Joegen

E. Baclor
Sent: Monday, April 09, 2007 5:19 AM
To: opensipstack-devel at lists dot sourceforge dot net
Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise

dev at whit dot ca wrote:

Members,

I''m doing some testing with the ATLSIP and opensipstack libraries
and so

far

with pretty good success. I have written a softphone in C# using the

samples

provided, however I have a strange issue which I think is related to

jitter

and/or comfort noise:

Setup:
C# Softphone ----> Asterisk ---> PRI -----> Telco

Once I make a call, the system works fine except if the person
using the
softphone talks for more then about 10-15 seconds (in a row without
being
interupted). Then, the audio starts to break up and the person on the

telco

side can''t make out what they are saying. Sometimes this situation is

reversed

and the person on the softphone can''t make out the person on the telco

side.

By the way, there aren''t any problems with the telco or asterisk
setup as

have

SIP hardphones using the system with no problems.

So my question is:

1. Can I send confort noise during silence breaks?

CNG is a codec functionality and is not manually generated by the
stack.

2. Where can I tweak the jitter-buffer or comfort noise settings?
Is this

done

SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed an the
ActiveX properties. Feel free to send in a patch if you get the chance
to expose it.

in the code itself?
3. Maybe I''m on the wrong track and any suggestions are welcome!

Look forward to working more with everyone on this exciting project!

Whit

Take Surveys. Earn Cash. Influence the Future of IT
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Guest

3. May 29, 2007 8:30 AM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Will prioritize this request. This should be available by tomorrow or on
early Thursday tops.

Regards,
Ilian

Joegen E. Baclor wrote:

Hi Ilian,

Can you provide an ETC for exposing Jitter and Silent Detection params
in ATLSIP? Seems like a popular request.

Joegen

Ilian Jeri C. Pinzon wrote:

Hi,

We haven''t exposed this yet but we will soon. Please wait for updates
in this list.

For the meantime, please refer to the attached email on how this can
be done.

Thanks.

Regards,
Ilian

Yacine Auczone wrote:

Hi All,
First, Thanks a lot for all the great job you are doing for
OpenSipStack and AtlSIP
I''m doing some devlopement test with the Softphone ActiveX, the
quality is very good and no bugs detected, the only thing is that the
softphone is doing by default some VAD and it is not transmiting the
silence, so there is no Comfort Noise generation sent whene the
calling party stop talking. i heard about a new ActiveX version which
will be available and gives the option to enable or disable CNG, is it
ready? if yes can i have it please?
Other Thing, on my Asterisk Server only G729 Work and not G729A
What''s Wrong ?

_______________________________________________
opensipstack-devel mailing list
opensipstack-devel at lists dot sourceforge dot net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

No virus found in this incoming message.
Checked by AVG Free Edition. Version: 7.5.472 / Virus Database:
269.8.1/822 - Release Date: 5/28/2007 11:40 AM

Subject:
[Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise]
From:
"Joegen E. Baclor"
Date:
Tue, 29 May 2007 18:26:21 +0800
To:
"Ilian Jeri C. Pinzon"

To:
"Ilian Jeri C. Pinzon"

Whit,

Good to hear you nailed it! Can''t wait to see your contributions if
you get the chance to expose the other setters/accessors in ATLSIP.

Whit Thiele wrote:

Whit

OpalSilenceDetector::Param param;
param.Mode = OpalSilenceDetector::NoSilenceDetection;
sfManager.SetSilenceDetectParams( params );

Hope that helps.

dev at whit dot ca wrote:

Joegen,

Thanks for the reply. I''ve been trying different jitterbuffer
settings as

well

as changing the number soundChannelBuffers to a number of different

settings

which I came across in some online
Opal documentation (

http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html

)

I''ve tried setting the jitter buffer to minimums 25 through to 500
and the

depth

to as high as 15 but nothing is helping. As I described before, I
can get

about

10-15 consecutive seconds of decent voice quality and then it gets very

choppy.

Is anyone else experiencing this?

I am wondering if it may have something to do with the Silence
detection

portion

of Opal. I''ve noticed in the opal.log file that the Silence Threshold

creeps

upwards the longer the person talks. Is there a way to disable the
silence
detector? I could see that there are several Modes (Fixed, Adaptive,
etc)

for

-----Original Message-----
From: opensipstack-devel-bounces at lists dot sourceforge dot net
[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On Behalf Of

Joegen

E. Baclor
Sent: Monday, April 09, 2007 5:19 AM
To: opensipstack-devel at lists dot sourceforge dot net
Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise

dev at whit dot ca wrote:

Members,

I''m doing some testing with the ATLSIP and opensipstack libraries
and so

far

with pretty good success. I have written a softphone in C# using the

samples

provided, however I have a strange issue which I think is related to

jitter

telco

side can''t make out what they are saying. Sometimes this situation is

reversed

and the person on the softphone can''t make out the person on the telco

side.

By the way, there aren''t any problems with the telco or asterisk
setup as

have

SIP hardphones using the system with no problems.

So my question is:

1. Can I send confort noise during silence breaks?

CNG is a codec functionality and is not manually generated by the
stack.

2. Where can I tweak the jitter-buffer or comfort noise settings?
Is this

done

SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed an the
ActiveX properties. Feel free to send in a patch if you get the chance
to expose it.

in the code itself?
3. Maybe I''m on the wrong track and any suggestions are welcome!

Look forward to working more with everyone on this exciting project!

Whit

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the chance to
share

your

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the chance to share

your

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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.472 / Virus Database: 269.8.1/822 - Release Date: 5/28/2007 11:40 AM

_______________________________________________
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https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Guest

4. May 30, 2007 5:53 AM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Hi all,

I have exposed the setting of silence detection mode and audio jitter
delay in ATLSIP and SoftPhoneInterface.

Here are the methods:

DisableSilenceDetection()

Disables silence detection. Disables CNG as well.

EnableFixedSilenceDetection( ULONG threshold )

Enables fixed silence detection. Any sound level below the threshold

is treated as silence (and CN is generated as a result). Don''t use too
high threshold values or you''ll only hear comfort noise. Try threshold=3
as suggested by Whit in another thread.

EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband,
ULONG silenceDeadband )

An extended version of the previous method. Don''t tinker with this

unless you know what you''re doing. For reference on how signalDeadband
and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket().

EnableAdaptiveSilenceDetection( ULONG adaptivePeriod )

Enables an adaptive silence detection. Supposedly this enables the

threshold to adapt to the current sound level every adaptivePeriod
milliseconds. However, its silence detection doesn''t seem to be very
effective (at least in my machine). I''ll look into this further to see
what''s wrong. This mode with adaptivePeriod=4800 is the default mode for
ATLSIP.

EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG
signalDeadband, ULONG silenceDeadband )

An extended version of the previous method. Don''t tinker with this

unless you know what you''re doing. For reference on how signalDeadband
and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket().

SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay )

Sets audio jitter delay settings.

Regards,
Ilian

Ilian Jeri C. Pinzon wrote:

Will prioritize this request. This should be available by tomorrow or on
early Thursday tops.

Regards,
Ilian

Joegen E. Baclor wrote:

Hi Ilian,

Can you provide an ETC for exposing Jitter and Silent Detection params
in ATLSIP? Seems like a popular request.

Joegen

Ilian Jeri C. Pinzon wrote:

Hi,

We haven''t exposed this yet but we will soon. Please wait for updates
in this list.

For the meantime, please refer to the attached email on how this can
be done.

Thanks.

Regards,
Ilian

Yacine Auczone wrote:

Hi All,
First, Thanks a lot for all the great job you are doing for
OpenSipStack and AtlSIP
I''m doing some devlopement test with the Softphone ActiveX, the
quality is very good and no bugs detected, the only thing is that the
softphone is doing by default some VAD and it is not transmiting the
silence, so there is no Comfort Noise generation sent whene the
calling party stop talking. i heard about a new ActiveX version which
will be available and gives the option to enable or disable CNG, is it
ready? if yes can i have it please?
Other Thing, on my Asterisk Server only G729 Work and not G729A
What''s Wrong ?

_______________________________________________
opensipstack-devel mailing list
opensipstack-devel at lists dot sourceforge dot net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

No virus found in this incoming message.
Checked by AVG Free Edition. Version: 7.5.472 / Virus Database:
269.8.1/822 - Release Date: 5/28/2007 11:40 AM

Subject:
[Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise]
From:
"Joegen E. Baclor"
Date:
Tue, 29 May 2007 18:26:21 +0800
To:
"Ilian Jeri C. Pinzon"

To:
"Ilian Jeri C. Pinzon"

Whit,

Good to hear you nailed it! Can''t wait to see your contributions if
you get the chance to expose the other setters/accessors in ATLSIP.

Whit Thiele wrote:

Whit

OpalSilenceDetector::Param param;
param.Mode = OpalSilenceDetector::NoSilenceDetection;
sfManager.SetSilenceDetectParams( params );

Hope that helps.

dev at whit dot ca wrote:

Joegen,

Thanks for the reply. I''ve been trying different jitterbuffer
settings as

well

as changing the number soundChannelBuffers to a number of different

settings

which I came across in some online
Opal documentation (

http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html

)

I''ve tried setting the jitter buffer to minimums 25 through to 500
and the

depth

to as high as 15 but nothing is helping. As I described before, I
can get

about

10-15 consecutive seconds of decent voice quality and then it gets very

choppy.

Is anyone else experiencing this?

I am wondering if it may have something to do with the Silence
detection

portion

of Opal. I''ve noticed in the opal.log file that the Silence Threshold

creeps

upwards the longer the person talks. Is there a way to disable the
silence
detector? I could see that there are several Modes (Fixed, Adaptive,
etc)

for

-----Original Message-----
From: opensipstack-devel-bounces at lists dot sourceforge dot net
[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On Behalf Of

Joegen

E. Baclor
Sent: Monday, April 09, 2007 5:19 AM
To: opensipstack-devel at lists dot sourceforge dot net
Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise

dev at whit dot ca wrote:

Members,

I''m doing some testing with the ATLSIP and opensipstack libraries
and so

far

with pretty good success. I have written a softphone in C# using the

samples

provided, however I have a strange issue which I think is related to

jitter

telco

side can''t make out what they are saying. Sometimes this situation is

reversed

and the person on the softphone can''t make out the person on the telco

side.

By the way, there aren''t any problems with the telco or asterisk
setup as

have

SIP hardphones using the system with no problems.

So my question is:

1. Can I send confort noise during silence breaks?

CNG is a codec functionality and is not manually generated by the
stack.

2. Where can I tweak the jitter-buffer or comfort noise settings?
Is this

done

SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed an the
ActiveX properties. Feel free to send in a patch if you get the chance
to expose it.

in the code itself?
3. Maybe I''m on the wrong track and any suggestions are welcome!

Look forward to working more with everyone on this exciting project!

Whit

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the chance to
share

your

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the chance to share

your

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the chance to share

your

No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.472 / Virus Database: 269.8.1/822 - Release Date: 5/28/2007 11:40 AM

_______________________________________________
opensipstack-devel mailing list
opensipstack-devel at lists dot sourceforge dot net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Guest

5. May 30, 2007 10:27 AM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Super! Thanks Ilian.

Ilian Jeri C. Pinzon wrote:

Hi all,

I have exposed the setting of silence detection mode and audio jitter
delay in ATLSIP and SoftPhoneInterface.

Here are the methods:

DisableSilenceDetection()

Disables silence detection. Disables CNG as well.

EnableFixedSilenceDetection( ULONG threshold )

Enables fixed silence detection. Any sound level below the threshold

An extended version of the previous method. Don''t tinker with this

Enables an adaptive silence detection. Supposedly this enables the

An extended version of the previous method. Don''t tinker with this

Sets audio jitter delay settings.

Regards,
Ilian

Ilian Jeri C. Pinzon wrote:

Will prioritize this request. This should be available by tomorrow or on
early Thursday tops.

Regards,
Ilian

Joegen E. Baclor wrote:

Hi Ilian,

Can you provide an ETC for exposing Jitter and Silent Detection params
in ATLSIP? Seems like a popular request.

Joegen

Ilian Jeri C. Pinzon wrote:

Hi,

We haven''t exposed this yet but we will soon. Please wait for updates
in this list.

For the meantime, please refer to the attached email on how this can
be done.

Thanks.

Regards,
Ilian

Yacine Auczone wrote:

Hi All,
First, Thanks a lot for all the great job you are doing for
OpenSipStack and AtlSIP
I''m doing some devlopement test with the Softphone ActiveX, the
quality is very good and no bugs detected, the only thing is that the
softphone is doing by default some VAD and it is not transmiting the
silence, so there is no Comfort Noise generation sent whene the
calling party stop talking. i heard about a new ActiveX version which
will be available and gives the option to enable or disable CNG, is it
ready? if yes can i have it please?
Other Thing, on my Asterisk Server only G729 Work and not G729A
What''s Wrong ?

_______________________________________________
opensipstack-devel mailing list
opensipstack-devel at lists dot sourceforge dot net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

No virus found in this incoming message.
Checked by AVG Free Edition. Version: 7.5.472 / Virus Database:
269.8.1/822 - Release Date: 5/28/2007 11:40 AM

Subject:
[Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise]
From:
"Joegen E. Baclor"
Date:
Tue, 29 May 2007 18:26:21 +0800
To:
"Ilian Jeri C. Pinzon"

To:
"Ilian Jeri C. Pinzon"

Whit,

Good to hear you nailed it! Can''t wait to see your contributions if
you get the chance to expose the other setters/accessors in ATLSIP.

Whit Thiele wrote:

Whit

OpalSilenceDetector::Param param;
param.Mode = OpalSilenceDetector::NoSilenceDetection;
sfManager.SetSilenceDetectParams( params );

Hope that helps.

dev at whit dot ca wrote:

Joegen,

Thanks for the reply. I''ve been trying different jitterbuffer
settings as

well

as changing the number soundChannelBuffers to a number of different

settings

which I came across in some online
Opal documentation (

http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html

)

I''ve tried setting the jitter buffer to minimums 25 through to 500
and the

depth

to as high as 15 but nothing is helping. As I described before, I
can get

about

10-15 consecutive seconds of decent voice quality and then it gets very

choppy.

Is anyone else experiencing this?

I am wondering if it may have something to do with the Silence
detection

portion

of Opal. I''ve noticed in the opal.log file that the Silence Threshold

creeps

upwards the longer the person talks. Is there a way to disable the
silence
detector? I could see that there are several Modes (Fixed, Adaptive,
etc)

for

-----Original Message-----
From: opensipstack-devel-bounces at lists dot sourceforge dot net
[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On Behalf Of

Joegen

E. Baclor
Sent: Monday, April 09, 2007 5:19 AM
To: opensipstack-devel at lists dot sourceforge dot net
Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise

dev at whit dot ca wrote:

Members,

I''m doing some testing with the ATLSIP and opensipstack libraries
and so

far

with pretty good success. I have written a softphone in C# using the

samples

provided, however I have a strange issue which I think is related to

jitter

telco

side can''t make out what they are saying. Sometimes this situation is

reversed

and the person on the softphone can''t make out the person on the telco

side.

By the way, there aren''t any problems with the telco or asterisk
setup as

have

SIP hardphones using the system with no problems.

So my question is:

1. Can I send confort noise during silence breaks?

CNG is a codec functionality and is not manually generated by the
stack.

2. Where can I tweak the jitter-buffer or comfort noise settings?
Is this

done

SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed an the
ActiveX properties. Feel free to send in a patch if you get the chance
to expose it.

in the code itself?
3. Maybe I''m on the wrong track and any suggestions are welcome!

Look forward to working more with everyone on this exciting project!

Whit

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Guest

6. Jun 2, 2007 9:35 AM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Hi,Thanks a lot for all your efforts.i have succesfully compiled ATLSIP with the new changes, but i have a little issue now.i''m not able to make calls since the updates, i''m getting a 403 Forbidden error code whene trying to make a call while i was able to make calls before.> Date: Wed, 30 May 2007 17:53:09 +0800> From: ipinzon at solegysystems dot com> To: opensipstack-devel at lists dot sourceforge dot net> Subject: Re: [OpenSIPStack] Cpmfort Noise Support> > Hi all,> > I have exposed the setting of silence detection mode and audio jitter > delay in ATLSIP and SoftPhoneInterface.> > Here are the methods:> > DisableSilenceDetection()> - Disables silence detection. Disables CNG as well.> > EnableFixedSilenceDetection( ULONG threshold )> - Enables fixed silence detection. Any sound level below the threshold > is treated as silence (and CN is generated as a result). Don''t use too > high threshold values or you''ll only hear comfort noise. Try threshold=3 > as suggested by Whit in another thread.> > EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband, > ULONG silenceDeadband )> - An extended version of the previous method. Don''t tinker with this > unless you know what you''re doing. For reference on how signalDeadband > and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket().> > EnableAdaptiveSilenceDetection( ULONG adaptivePeriod )> - Enables an adaptive silence detection. Supposedly this enables the > threshold to adapt to the current sound level every adaptivePeriod > milliseconds. However, its silence detection doesn''t seem to be very > effective (at least in my machine). I''ll look into this further to see > what''s wrong. This mode with adaptivePeriod=4800 is the default mode for > ATLSIP.> > EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG > signalDeadband, ULONG silenceDeadband )> - An extended version of the previous method. Don''t tinker with this > unless you know what you''re doing. For reference on how signalDeadband > and silenceDeadband are used, look in OpalSilenceDetector::ReceivedPacket().> > SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay )> - Sets audio jitter delay settings.> > > Regards,> Ilian> > Ilian Jeri C. Pinzon wrote:> > Will prioritize this request. This should be available by tomorrow or on > > early Thursday tops.> >> > Regards,> > Ilian> >> > Joegen E. Baclor wrote:> > > >> Hi Ilian,> >>> >> Can you provide an ETC for exposing Jitter and Silent Detection params > >> in ATLSIP? Seems like a popular request.> >>> >> Joegen> >>> >>> >> Ilian Jeri C. Pinzon wrote:> >> > >> > >>> Hi,> >>>> >>> We haven''t exposed this yet but we will soon. Please wait for updates > >>> in this list.> >>>> >>> For the meantime, please refer to the attached email on how this can > >>> be done.> >>>> >>> Thanks.> >>>> >>> Regards,> >>> Ilian> >>>> >>> Yacine Auczone wrote:> >>> > >>> > >>>> Hi All,> >>>> First, Thanks a lot for all the great job you are doing for> >>>> OpenSipStack and AtlSIP> >>>> I''m doing some devlopement test with the Softphone ActiveX, the> >>>> quality is very good and no bugs detected, the only thing is that the> >>>> softphone is doing by default some VAD and it is not transmiting the> >>>> silence, so there is no Comfort Noise generation sent whene the> >>>> calling party stop talking. i heard about a new ActiveX version which> >>>> will be available and gives the option to enable or disable CNG, is it> >>>> ready? if yes can i have it please?> >>>> Other Thing, on my Asterisk Server only G729 Work and not G729A> >>>> What''s Wrong ?> >>>>> >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------------> >>>> Avec Windows Live Spaces, publiez directement des messages > >>>> électroniques sur votre blog ou ajoutez-y des photos, des blagues et > >>>> d''autres infos. C''est gratuit ! > >>>> <http://clk.atdmt.com/MSN/go/msnnksac0030000001msn/direct/01/?href=http://www.imagine-msn.com/spaces> > >>>>> >>>> ------------------------------------------------------------------------> >>>>> >>>>

> >>>>> >>>> This SF.net email is sponsored by DB2 Express> >>>> Download DB2 Express C - the FREE version of DB2 express and take> >>>> control of your XML. No limits. Just data. Click to get it now.> >>>> http://sourceforge.net/powerbar/db2/> >>>> ------------------------------------------------------------------------> >>>>> >>>> _______________________________________________> >>>> opensipstack-devel mailing list> >>>> opensipstack-devel at lists dot sourceforge dot net> >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >>>> > >>>> ------------------------------------------------------------------------> >>>>> >>>> No virus found in this incoming message.> >>>> Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: > >>>> 269.8.1/822 - Release Date: 5/28/2007 11:40 AM> >>>> > >>>> > >>>> > >>> ------------------------------------------------------------------------> >>>> >>> Subject:> >>> [Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise]> >>> From:> >>> "Joegen E. Baclor" > >>> Date:> >>> Tue, 29 May 2007 18:26:21 +0800> >>> To:> >>> "Ilian Jeri C. Pinzon" > >>>> >>> To:> >>> "Ilian Jeri C. Pinzon" > >>>> >>>> >>>> >>>> >>> ------------------------------------------------------------------------> >>>> >>> Subject:> >>> Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise> >>> From:> >>> "Joegen E. Baclor" > >>> Date:> >>> Thu, 12 Apr 2007 16:40:04 +0800> >>> To:> >>> opensipstack-devel at lists dot sourceforge dot net> >>>> >>> To:> >>> opensipstack-devel at lists dot sourceforge dot net> >>>> >>>> >>> Whit,> >>>> >>> Good to hear you nailed it! Can''t wait to see your contributions if > >>> you get the chance to expose the other setters/accessors in ATLSIP.> >>>> >>>> >>>> >>> Whit Thiele wrote:> >>> > >>> > >>>> Joegen,> >>>>> >>>> Thanks for the help. I thought I''d send the list an update on what > >>>> solved my> >>>> problem. I changed the Silence Detector to Fixed with a threshold of > >>>> 3. This> >>>> eliminated all the problems! It seems that the adaptive silence > >>>> detector was> >>>> constantly incrementing and started affecting things about 10-15 seconds> >>>> into a conversation!> >>>>> >>>> I''ll probably put in the ability to change the jitterbuffer and silence> >>>> detector into the ATLSIP library and send this in to the project in > >>>> the next> >>>> couple weeks...> >>>>> >>>>> >>>> Whit> >>>>> >>>>> >>>>> >>>> -----Original Message-----> >>>> From: opensipstack-devel-bounces at lists dot sourceforge dot net> >>>> [mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On Behalf Of> >>>> Joegen E. Baclor> >>>> Sent: Tuesday, April 10, 2007 9:36 PM> >>>> To: opensipstack-devel at lists dot sourceforge dot net> >>>> Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise> >>>>> >>>> Whit,> >>>>> >>>> It is probably best to ask this question to > >>>> openh323-devel at lists dot sourceforge dot net. However, here''s how to set > >>>> the silence detection in code.> >>>>> >>>>> >>>>> >>>> OpalSilenceDetector::Param param;> >>>> param.Mode = OpalSilenceDetector::NoSilenceDetection;> >>>> sfManager.SetSilenceDetectParams( params );> >>>>> >>>>> >>>>> >>>>> >>>> Hope that helps.> >>>>> >>>> dev at whit dot ca wrote:> >>>> > >>>> > >>>> > >>>>> Joegen,> >>>>>> >>>>> Thanks for the reply. I''ve been trying different jitterbuffer > >>>>> settings as> >>>>> > >>>>> > >>>>> > >>>> well> >>>> > >>>> > >>>> > >>>>> as changing the number soundChannelBuffers to a number of different> >>>>> > >>>>> > >>>>> > >>>> settings> >>>> > >>>> > >>>> > >>>>> which I came across in some online> >>>>> Opal documentation (> >>>>>> >>>>> > >>>>> > >>>>> > >>>> http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html > >>>>> >>>> )> >>>> > >>>> > >>>> > >>>>> I''ve tried setting the jitter buffer to minimums 25 through to 500 > >>>>> and the> >>>>> > >>>>> > >>>>> > >>>> depth> >>>> > >>>> > >>>> > >>>>> to as high as 15 but nothing is helping. As I described before, I > >>>>> can get> >>>>> > >>>>> > >>>>> > >>>> about> >>>> > >>>> > >>>> > >>>>> 10-15 consecutive seconds of decent voice quality and then it gets very> >>>>> > >>>>> > >>>>> > >>>> choppy. > >>>> > >>>> > >>>>> Is anyone else experiencing this?> >>>>>> >>>>> I am wondering if it may have something to do with the Silence > >>>>> detection> >>>>> > >>>>> > >>>>> > >>>> portion> >>>> > >>>> > >>>> > >>>>> of Opal. I''ve noticed in the opal.log file that the Silence Threshold> >>>>> > >>>>> > >>>>> > >>>> creeps> >>>> > >>>> > >>>> > >>>>> upwards the longer the person talks. Is there a way to disable the > >>>>> silence> >>>>> detector? I could see that there are several Modes (Fixed, Adaptive, > >>>>> etc)> >>>>> > >>>>> > >>>>> > >>>> for> >>>> > >>>> > >>>> > >>>>> it but I can''t figure out where this is initialized in the code.> >>>>> I may be on the wrong track but I can''t figure out this strange > >>>>> behavior.> >>>>> Any help/ideas/suggestions would be greatly appreciated!> >>>>>> >>>>> Whit> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> -----Original Message-----> >>>>> From: opensipstack-devel-bounces at lists dot sourceforge dot net> >>>>> [mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On Behalf Of> >>>>> > >>>>> > >>>>> > >>>> Joegen> >>>> > >>>> > >>>> > >>>>> E. Baclor> >>>>> Sent: Monday, April 09, 2007 5:19 AM> >>>>> To: opensipstack-devel at lists dot sourceforge dot net> >>>>> Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise> >>>>>> >>>>> dev at whit dot ca wrote:> >>>>> > >>>>> > >>>>> > >>>>>> Members,> >>>>>>> >>>>>> I''m doing some testing with the ATLSIP and opensipstack libraries > >>>>>> and so> >>>>>> > >>>>>> > >>>>>> > >>>> far> >>>> > >>>> > >>>> > >>>>>> with pretty good success. I have written a softphone in C# using the> >>>>>> > >>>>>> > >>>>>> > >>>> samples> >>>> > >>>> > >>>> > >>>>>> provided, however I have a strange issue which I think is related to> >>>>>> > >>>>>> > >>>>>> > >>>> jitter> >>>> > >>>> > >>>> > >>>>>> and/or comfort noise:> >>>>>>> >>>>>> Setup:> >>>>>> C# Softphone ----> Asterisk ---> PRI -----> Telco> >>>>>>> >>>>>> Once I make a call, the system works fine except if the person > >>>>>> using the> >>>>>> softphone talks for more then about 10-15 seconds (in a row without > >>>>>> being> >>>>>> interupted). Then, the audio starts to break up and the person on the> >>>>>> > >>>>>> > >>>>>> > >>>> telco> >>>> > >>>> > >>>> > >>>>>> side can''t make out what they are saying. Sometimes this situation is> >>>>>> > >>>>>> > >>>>>> > >>>> reversed> >>>> > >>>> > >>>> > >>>>>> and the person on the softphone can''t make out the person on the telco> >>>>>> > >>>>>> > >>>>>> > >>>> side. > >>>> > >>>> > >>>>>> By the way, there aren''t any problems with the telco or asterisk > >>>>>> setup as> >>>>>> > >>>>>> > >>>>>> > >>>> I> >>>> > >>>> > >>>> > >>>>>> > >>>>>> > >>>>>> > >>>>> have> >>>>> > >>>>> > >>>>> > >>>>>> SIP hardphones using the system with no problems.> >>>>>>> >>>>>> So my question is:> >>>>>>> >>>>>>> >>>>>> 1. Can I send confort noise during silence breaks?> >>>>>>> >>>>>> > >>>>>> > >>>>>> > >>>>> CNG is a codec functionality and is not manually generated by the > >>>>> stack.> >>>>>> >>>>>> >>>>> > >>>>> > >>>>> > >>>>>> 2. Where can I tweak the jitter-buffer or comfort noise settings? > >>>>>> Is this> >>>>>> > >>>>>> > >>>>>> > >>>> done> >>>> > >>>> > >>>> > >>>>>> > >>>>>> > >>>>>> > >>>>> SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed an the> >>>>> ActiveX properties. Feel free to send in a patch if you get the chance> >>>>> to expose it.> >>>>>> >>>>> > >>>>> > >>>>> > >>>>>> in the code itself?> >>>>>> 3. Maybe I''m on the wrong track and any suggestions are welcome!> >>>>>>> >>>>>>> >>>>>> Look forward to working more with everyone on this exciting project!> >>>>>>> >>>>>> Whit> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>

> >>>>>>> >>>>>> Take Surveys. Earn Cash. Influence the Future of IT> >>>>>> Join SourceForge.net''s Techsay panel and you''ll get the chance to > >>>>>> share> >>>>>> > >>>>>> > >>>>>> > >>>> your> >>>> > >>>> > >>>> > >>>>>> opinions on IT & business topics through brief surveys-and earn cash> >>>>>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > >>>>>>> >>>>>> _______________________________________________> >>>>>> opensipstack-devel mailing list> >>>>>> opensipstack-devel at lists dot sourceforge dot net> >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >>>>>>> >>>>>>> >>>>>> > >>>>>> > >>>>>> > >>>>>

> >>>>>> >>>>> Take Surveys. Earn Cash. Influence the Future of IT> >>>>> Join SourceForge.net''s Techsay panel and you''ll get the chance to share> >>>>> > >>>>> > >>>>> > >>>> your> >>>> > >>>> > >>>> > >>>>> opinions on IT & business topics through brief surveys-and earn cash> >>>>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > >>>>>> >>>>> _______________________________________________> >>>>> opensipstack-devel mailing list> >>>>> opensipstack-devel at lists dot sourceforge dot net> >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >>>>>> >>>>>> >>>>>

> >>>>> >>>> Take Surveys. Earn Cash. Influence the Future of IT> >>>> Join SourceForge.net''s Techsay panel and you''ll get the chance to > >>>> share your> >>>> opinions on IT & business topics through brief surveys-and earn cash> >>>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > >>>>> >>>> _______________________________________________> >>>> opensipstack-devel mailing list> >>>> opensipstack-devel at lists dot sourceforge dot net> >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >>>>> >>>>> >>>> > >>>> > >>>> > >>> ------------------------------------------------------------------------> >>>> >>> No virus found in this incoming message.> >>> Checked by AVG Free Edition. > >>> Version: 7.5.472 / Virus Database: 269.8.1/822 - Release Date: 5/28/2007 11:40 AM> >>> > >>> ------------------------------------------------------------------------> >>>> >>> -------------------------------------------------------------------------> >>> This SF.net email is sponsored by DB2 Express> >>> Download DB2 Express C - the FREE version of DB2 express and take> >>> control of your XML. No limits. Just data. Click to get it now.> >>> http://sourceforge.net/powerbar/db2/> >>> ------------------------------------------------------------------------> >>>> >>> _______________________________________________> >>> opensipstack-devel mailing list> >>> opensipstack-devel at lists dot sourceforge dot net> >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >>> > >>> > >>> > >> -------------------------------------------------------------------------> >> This SF.net email is sponsored by DB2 Express> >> Download DB2 Express C - the FREE version of DB2 express and take> >> control of your XML. No limits. Just data. Click to get it now.> >> http://sourceforge.net/powerbar/db2/> >> _______________________________________________> >> opensipstack-devel mailing list> >> opensipstack-devel at lists dot sourceforge dot net> >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >>> >>> >> > >> > >> >> > -------------------------------------------------------------------------> > This SF.net email is sponsored by DB2 Express> > Download DB2 Express C - the FREE version of DB2 express and take> > control of your XML. No limits. Just data. Click to get it now.> > http://sourceforge.net/powerbar/db2/> > _______________________________________________> > opensipstack-devel mailing list> > opensipstack-devel at lists dot sourceforge dot net> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel> >> >> > > > > -------------------------------------------------------------------------> This SF.net email is sponsored by DB2 Express> Download DB2 Express C - the FREE version of DB2 express and take> control of your XML. No limits. Just data. Click to get it now.> http://sourceforge.net/powerbar/db2/> _______________________________________________> opensipstack-devel mailing list> opensipstack-devel at lists dot sourceforge dot net> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
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Guest

7. Jun 2, 2007 11:26 PM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Hi Yacine,

Please send Ilian a level sip 5 log so he can determine the casue and
give you a fix. Thanks.

Joegen

Yacine Auczone wrote:

Date: Wed, 30 May 2007 17:53:09 +0800
From: ipinzon at solegysystems dot com
To: opensipstack-devel at lists dot sourceforge dot net
Subject: Re: [OpenSIPStack] Cpmfort Noise Support

Hi all,

I have exposed the setting of silence detection mode and audio jitter
delay in ATLSIP and SoftPhoneInterface.

Here are the methods:

DisableSilenceDetection()

Disables silence detection. Disables CNG as well.

EnableFixedSilenceDetection( ULONG threshold )

Enables fixed silence detection. Any sound level below the threshold

is treated as silence (and CN is generated as a result). Don''t use too
high threshold values or you''ll only hear comfort noise. Try

threshold=3

as suggested by Whit in another thread.

EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband,
ULONG silenceDeadband )

An extended version of the previous method. Don''t tinker with this

unless you know what you''re doing. For reference on how signalDeadband
and silenceDeadband are used, look in

OpalSilenceDetector::ReceivedPacket().

EnableAdaptiveSilenceDetection( ULONG adaptivePeriod )

Enables an adaptive silence detection. Supposedly this enables the

for

ATLSIP.

EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG
signalDeadband, ULONG silenceDeadband )

An extended version of the previous method. Don''t tinker with this

unless you know what you''re doing. For reference on how signalDeadband
and silenceDeadband are used, look in

OpalSilenceDetector::ReceivedPacket().

SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay )

Sets audio jitter delay settings.

Regards,
Ilian

Ilian Jeri C. Pinzon wrote:

Will prioritize this request. This should be available by tomorrow

or on

early Thursday tops.

Regards,
Ilian

Joegen E. Baclor wrote:

Hi Ilian,

Can you provide an ETC for exposing Jitter and Silent Detection

params

in ATLSIP? Seems like a popular request.

Joegen

Ilian Jeri C. Pinzon wrote:

Hi,

We haven''t exposed this yet but we will soon. Please wait for

updates

in this list.

For the meantime, please refer to the attached email on how this

can

be done.

Thanks.

Regards,
Ilian

Yacine Auczone wrote:

that the

softphone is doing by default some VAD and it is not

transmiting the

silence, so there is no Comfort Noise generation sent whene the
calling party stop talking. i heard about a new ActiveX version

which

will be available and gives the option to enable or disable

CNG, is it

ready? if yes can i have it please?
Other Thing, on my Asterisk Server only G729 Work and not G729A
What''s Wrong ?

Avec Windows Live Spaces, publiez directement des messages
électroniques sur votre blog ou ajoutez-y des photos, des

blagues et

d''autres infos. C''est gratuit !

<http://clk.atdmt.com/MSN/go/msnnksac0030000001msn/direct/01/?href=http://www.imagine-msn.com/spaces>

_______________________________________________
opensipstack-devel mailing list
opensipstack-devel at lists dot sourceforge dot net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

No virus found in this incoming message.
Checked by AVG Free Edition. Version: 7.5.472 / Virus Database:
269.8.1/822 - Release Date: 5/28/2007 11:40 AM

Subject:
[Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise]
From:
"Joegen E. Baclor"
Date:
Tue, 29 May 2007 18:26:21 +0800
To:
"Ilian Jeri C. Pinzon"

To:
"Ilian Jeri C. Pinzon"

Whit,

Good to hear you nailed it! Can''t wait to see your contributions if
you get the chance to expose the other setters/accessors in ATLSIP.

Whit Thiele wrote:

Joegen,

Thanks for the help. I thought I''d send the list an update on what
solved my
problem. I changed the Silence Detector to Fixed with a

threshold of

3. This
eliminated all the problems! It seems that the adaptive silence
detector was
constantly incrementing and started affecting things about

10-15 seconds

into a conversation!

I''ll probably put in the ability to change the jitterbuffer and

silence

detector into the ATLSIP library and send this in to the

project in

the next
couple weeks...

Whit

-----Original Message-----
From: opensipstack-devel-bounces at lists dot sourceforge dot net
[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On

Behalf Of

Joegen E. Baclor
Sent: Tuesday, April 10, 2007 9:36 PM
To: opensipstack-devel at lists dot sourceforge dot net
Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort

Noise

Whit,

It is probably best to ask this question to
openh323-devel at lists dot sourceforge dot net. However, here''s how to set
the silence detection in code.

OpalSilenceDetector::Param param;
param.Mode = OpalSilenceDetector::NoSilenceDetection;
sfManager.SetSilenceDetectParams( params );

Hope that helps.

dev at whit dot ca wrote:

Joegen,

Thanks for the reply. I''ve been trying different jitterbuffer
settings as

well

as changing the number soundChannelBuffers to a number of

different

settings

which I came across in some online
Opal documentation (

http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html

)

I''ve tried setting the jitter buffer to minimums 25 through to

500

and the

depth

to as high as 15 but nothing is helping. As I described before, I
can get

about

10-15 consecutive seconds of decent voice quality and then it

gets very

choppy.

Is anyone else experiencing this?

I am wondering if it may have something to do with the Silence
detection

portion

of Opal. I''ve noticed in the opal.log file that the Silence

Threshold

creeps

upwards the longer the person talks. Is there a way to disable

the

silence
detector? I could see that there are several Modes (Fixed,

Adaptive,

etc)

for

-----Original Message-----
From: opensipstack-devel-bounces at lists dot sourceforge dot net
[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On

Behalf Of

Joegen

E. Baclor
Sent: Monday, April 09, 2007 5:19 AM
To: opensipstack-devel at lists dot sourceforge dot net
Subject: Re: [OpenSIPStack] Audio problems - Jitter and

Comfort Noise

dev at whit dot ca wrote:

Members,

I''m doing some testing with the ATLSIP and opensipstack

libraries

and so

far

with pretty good success. I have written a softphone in C#

using the

samples

provided, however I have a strange issue which I think is

related to

jitter

without

being
interupted). Then, the audio starts to break up and the

person on the

telco

side can''t make out what they are saying. Sometimes this

situation is

reversed

and the person on the softphone can''t make out the person on

the telco

side.

By the way, there aren''t any problems with the telco or asterisk
setup as

have

SIP hardphones using the system with no problems.

So my question is:

1. Can I send confort noise during silence breaks?

CNG is a codec functionality and is not manually generated by the
stack.

2. Where can I tweak the jitter-buffer or comfort noise

settings?

Is this

done

SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed

an the

ActiveX properties. Feel free to send in a patch if you get

the chance

to expose it.

in the code itself?
3. Maybe I''m on the wrong track and any suggestions are welcome!

Look forward to working more with everyone on this exciting

project!

Whit

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the

chance to

your

opinions on IT & business topics through brief surveys-and

earn cash

http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV

_______________________________________________
opensipstack-devel mailing list
opensipstack-devel at lists dot sourceforge dot net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the chance

to share

your

opinions on IT & business topics through brief surveys-and

earn cash

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_______________________________________________
opensipstack-devel mailing list
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https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

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Join SourceForge.net''s Techsay panel and you''ll get the chance

to share

your

opinions on IT & business topics through brief surveys-and

earn cash

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_______________________________________________
opensipstack-devel mailing list
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https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the chance to
share your
opinions on IT & business topics through brief surveys-and earn

cash

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https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

Take Surveys. Earn Cash. Influence the Future of IT
Join SourceForge.net''s Techsay panel and you''ll get the chance to
share your
opinions on IT & business topics through brief surveys-and earn

cash

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opensipstack-devel at lists dot sourceforge dot net
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.472 / Virus Database: 269.8.1/822 - Release Date:

5/28/2007 11:40 AM

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Guest

8. Jun 3, 2007 5:06 AM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Hi,

This is the log i''m getting

----------------16:09.322----------------

LISTENER STARTED *** 127.0.0.1:5060

----------------16:09.513----------------

LISTENER STARTED *** 192.168.0.53:5060 [*** DEFAULT LISTENER ***]

----------------16:09.612----------------

SEND: enc=0 546 Bytes to 193.194.64.11:5060:UDP (REGISTER sip:193.194.64.11 SIP/2.0) Interface Address=

REGISTER sip:193.194.64.11 SIP/2.0

From: 7000 ;tag=598b5f0dd6f8181099ef9f8b648defb9

To: sip:7000@193.194.64.11

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK74915f0dd6f8181099ef9f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 1 REGISTER

Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9

Contact: "7000" <sip:7000@192.168.0.53:5060;transport=udp>

User-Agent: OpenSIPStack-1.1.6-168

Expires: 3600

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:09.812----------------

RCV: enc=0 493 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 100 Trying)

SIP/2.0 100 Trying

From: 7000 ;tag=598b5f0dd6f8181099ef9f8b648defb9

To: sip:7000@193.194.64.11

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK74915f0dd6f8181099ef9f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 1 REGISTER

Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9

Contact:

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:09.835----------------

RCV: enc=0 553 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 401 Unauthorized)

SIP/2.0 401 Unauthorized

From: 7000 ;tag=598b5f0dd6f8181099ef9f8b648defb9

To: sip:7000@193.194.64.11;tag=as186cc90c

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK74915f0dd6f8181099ef9f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 1 REGISTER

Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9

User-Agent: Asterisk PBX

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="549f3188"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:09.899----------------

SEND: enc=0 708 Bytes to 193.194.64.11:5060:UDP (REGISTER sip:193.194.64.11 SIP/2.0) Interface Address=

REGISTER sip:193.194.64.11 SIP/2.0

From: 7000 ;tag=51ff5f0dd6f8181099ef9f8b648defb9

To: sip:7000@193.194.64.11

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK51ff5f0dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 2 REGISTER

Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9

Contact: "7000" <sip:7000@192.168.0.53:5060;transport=udp>

User-Agent: OpenSIPStack-1.1.6-168

Expires: 3600

Max-Forwards: 10

Authorization: Digest username="7000", realm="asterisk", nonce="549f3188", uri="sip:193.194.64.11", response="ae2f9b140c8cd0c80f6f0522cf29364f", algorithm=MD5

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:09.917----------------

RCV: enc=0 493 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 100 Trying)

SIP/2.0 100 Trying

From: 7000 ;tag=51ff5f0dd6f8181099ef9f8b648defb9

To: sip:7000@193.194.64.11

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK51ff5f0dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 2 REGISTER

Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9

Contact:

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:09.925----------------

RCV: enc=0 585 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 200 OK)

SIP/2.0 200 OK

From: 7000 ;tag=51ff5f0dd6f8181099ef9f8b648defb9

To: sip:7000@193.194.64.11;tag=as186cc90c

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4033;branch=z9hG4bK51ff5f0dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 2 REGISTER

Call-ID: 598b5f0d-d6f8-1810-8b21-9f8b648defb9

Contact: <sip:7000@192.168.0.53:5060;transport=udp>;expires=3600

Date: Sun, 03 Jun 2007 09:31:56 GMT

User-Agent: Asterisk PBX

Expires: 3600

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:22.831----------------

SEND: enc=0 754 Bytes to 193.194.64.11:5060:UDP (INVITE sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

INVITE sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4711 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Type: application/sdp

Content-Length: 203

v=0

o=- 1180863134 1180863134 IN IP4 192.168.0.53

s=OSS RTP Session

c=IN IP4 192.168.0.53

t=0 0

m=audio 5000 RTP/AVP 101 8

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:8 PCMA/8000

----------------16:22.858----------------

RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required)

SIP/2.0 407 Proxy Authentication Required

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 4711 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

User-Agent: Asterisk PBX

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:22.908----------------

SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

ACK sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4711 ACK

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:22.961----------------

SEND: enc=0 927 Bytes to 193.194.64.11:5060:UDP (INVITE sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

INVITE sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKdbed730dd6f8181099f19f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4712 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Proxy-Authorization: Digest username="7000", realm="asterisk", nonce="797c48aa", uri="sip:5000@193.194.64.11", response="98b951dc98b2ae41bfa445f6225af411", algorithm=MD5

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Type: application/sdp

Content-Length: 203

----------------16:22.977----------------

RCV: enc=0 472 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 403 Forbidden)

SIP/2.0 403 Forbidden

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKdbed730dd6f8181099f19f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 4712 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:22.992----------------

SEND: enc=0 700 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

ACK sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKdbed730dd6f8181099f19f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4712 ACK

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Proxy-Authorization: Digest username="7000", realm="asterisk", nonce="797c48aa", uri="sip:5000@193.194.64.11", response="98b951dc98b2ae41bfa445f6225af411", algorithm=MD5

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:23.858----------------

RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required)

SIP/2.0 407 Proxy Authentication Required

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 4711 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

User-Agent: Asterisk PBX

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:23.888----------------

SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

ACK sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4711 ACK

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:24.859----------------

RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required)

SIP/2.0 407 Proxy Authentication Required

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 4711 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

User-Agent: Asterisk PBX

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:24.889----------------

SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

ACK sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4711 ACK

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:26.860----------------

RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required)

SIP/2.0 407 Proxy Authentication Required

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 4711 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

User-Agent: Asterisk PBX

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:26.893----------------

SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

ACK sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4711 ACK

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:30.859----------------

RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required)

SIP/2.0 407 Proxy Authentication Required

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 4711 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

User-Agent: Asterisk PBX

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:30.895----------------

SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

ACK sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4711 ACK

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:34.860----------------

RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required)

SIP/2.0 407 Proxy Authentication Required

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 4711 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

User-Agent: Asterisk PBX

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:34.893----------------

SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

ACK sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4711 ACK

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

----------------16:38.862----------------

RCV: enc=0 570 Bytes from RCVADDR: 193.194.64.11:RCVPORT: 5060:UDP (SIP/2.0 407 Proxy Authentication Required)

SIP/2.0 407 Proxy Authentication Required

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport=5060;received=192.168.0.53

CSeq: 4711 INVITE

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

User-Agent: Asterisk PBX

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="797c48aa"

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0

----------------16:38.891----------------

SEND: enc=0 527 Bytes to 193.194.64.11:5060:UDP (ACK sip:5000@193.194.64.11 SIP/2.0) Interface Address=192.168.0.53

ACK sip:5000@193.194.64.11 SIP/2.0

From: 7000 ;tag=c973730dd6f8181099f19f8b648defb9

To: sip:5000@193.194.64.11;tag=as318d4032

Via: SIP/2.0/UDP 192.168.0.53:5060;iid=4034;branch=z9hG4bKc973730dd6f8181099f09f8b648defb9;uas-addr=193.194.64.11;rport

CSeq: 4711 ACK

Call-ID: c973730d-d6f8-1810-8b22-9f8b648defb9

Contact: "7000"

User-Agent: OpenSIPStack-1.1.6-168

Max-Forwards: 10

Allow: INVITE, BYE, ACK, REFER, MESSAGE, INFO, NOTIFY, OPTIONS

Content-Length: 0

Thanks

On Sun, 03 Jun 2007 11:26:45 +0800, "Joegen E. Baclor" wrote:

Hi Yacine,

Please send Ilian a level sip 5 log so he can determine the casue and

give you a fix. Thanks.

Joegen

Yacine Auczone wrote:

Hi,

Thanks a lot for all your efforts.

i have succesfully compiled ATLSIP with the new changes, but i have a

little issue now.

i''m not able to make calls since the updates, i''m getting a 403

Forbidden error code whene trying to make a call while i was able to

make calls before.

------------------------------------------------------------------------

Date: Wed, 30 May 2007 17:53:09 +0800

From: ipinzon at solegysystems dot com

To: opensipstack-devel at lists dot sourceforge dot net

Subject: Re: [OpenSIPStack] Cpmfort Noise Support

Hi all,

I have exposed the setting of silence detection mode and audio jitter

delay in ATLSIP and SoftPhoneInterface.

Here are the methods:

DisableSilenceDetection()

- Disables silence detection. Disables CNG as well.

EnableFixedSilenceDetection( ULONG threshold )

- Enables fixed silence detection. Any sound level below the threshold

is treated as silence (and CN is generated as a result). Don''t use too

high threshold values or you''ll only hear comfort noise. Try

threshold=3

as suggested by Whit in another thread.

EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband,

ULONG silenceDeadband )

- An extended version of the previous method. Don''t tinker with this

unless you know what you''re doing. For reference on how signalDeadband

and silenceDeadband are used, look in

OpalSilenceDetector::ReceivedPacket().

EnableAdaptiveSilenceDetection( ULONG adaptivePeriod )

- Enables an adaptive silence detection. Supposedly this enables the

threshold to adapt to the current sound level every adaptivePeriod

milliseconds. However, its silence detection doesn''t seem to be very

effective (at least in my machine). I''ll look into this further to see

what''s wrong. This mode with adaptivePeriod=4800 is the default mode

for

ATLSIP.

EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG

signalDeadband, ULONG silenceDeadband )

- An extended version of the previous method. Don''t tinker with this

unless you know what you''re doing. For reference on how signalDeadband

and silenceDeadband are used, look in

OpalSilenceDetector::ReceivedPacket().

SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay )

- Sets audio jitter delay settings.

Regards,

Ilian

Ilian Jeri C. Pinzon wrote:

Will prioritize this request. This should be available by tomorrow

or on

early Thursday tops.

Regards,

Ilian

Joegen E. Baclor wrote:

Hi Ilian,

Can you provide an ETC for exposing Jitter and Silent Detection

params

in ATLSIP? Seems like a popular request.

Joegen

Ilian Jeri C. Pinzon wrote:

Hi,

We haven''t exposed this yet but we will soon. Please wait for

updates

in this list.

For the meantime, please refer to the attached email on how this

can

be done.

Thanks.

Regards,

Ilian

Yacine Auczone wrote:

Hi All,

First, Thanks a lot for all the great job you are doing for

OpenSipStack and AtlSIP

I''m doing some devlopement test with the Softphone ActiveX, the

quality is very good and no bugs detected, the only thing is

that the

softphone is doing by default some VAD and it is not

transmiting the

silence, so there is no Comfort Noise generation sent whene the

calling party stop talking. i heard about a new ActiveX version

which

will be available and gives the option to enable or disable

CNG, is it

ready? if yes can i have it please?

Other Thing, on my Asterisk Server only G729 Work and not G729A

What''s Wrong ?

------------------------------------------------------------------------

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Subject:

[Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort

Noise]

From:

"Joegen E. Baclor"

Date:

Tue, 29 May 2007 18:26:21 +0800

To:

"Ilian Jeri C. Pinzon"

To:

"Ilian Jeri C. Pinzon"

------------------------------------------------------------------------

Subject:

Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise

From:

"Joegen E. Baclor"

Date:

Thu, 12 Apr 2007 16:40:04 +0800

To:

opensipstack-devel at lists dot sourceforge dot net

To:

opensipstack-devel at lists dot sourceforge dot net

Whit,

Good to hear you nailed it! Can''t wait to see your contributions

you get the chance to expose the other setters/accessors in

ATLSIP.

Whit Thiele wrote:

Joegen,

Thanks for the help. I thought I''d send the list an update on

what

solved my

problem. I changed the Silence Detector to Fixed with a

threshold of

3. This

eliminated all the problems! It seems that the adaptive silence

detector was

constantly incrementing and started affecting things about

10-15 seconds

into a conversation!

I''ll probably put in the ability to change the jitterbuffer and

silence

detector into the ATLSIP library and send this in to the

project in

the next

couple weeks...

Whit

-----Original Message-----

From: opensipstack-devel-bounces at lists dot sourceforge dot net

[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On

Behalf Of

Joegen E. Baclor

Sent: Tuesday, April 10, 2007 9:36 PM

To: opensipstack-devel at lists dot sourceforge dot net

Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort

Noise

Whit,

It is probably best to ask this question to

openh323-devel at lists dot sourceforge dot net. However, here''s how to set

the silence detection in code.

OpalSilenceDetector::Param param;

param.Mode = OpalSilenceDetector::NoSilenceDetection;

sfManager.SetSilenceDetectParams( params );

Hope that helps.

dev at whit dot ca wrote:

Joegen,

Thanks for the reply. I''ve been trying different jitterbuffer

settings as

well

as changing the number soundChannelBuffers to a number of

different

settings

which I came across in some online

Opal documentation (

http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html

)

I''ve tried setting the jitter buffer to minimums 25 through to

500

and the

depth

to as high as 15 but nothing is helping. As I described before,

can get

about

10-15 consecutive seconds of decent voice quality and then it

gets very

choppy.

Is anyone else experiencing this?

I am wondering if it may have something to do with the Silence

detection

portion

of Opal. I''ve noticed in the opal.log file that the Silence

Threshold

creeps

upwards the longer the person talks. Is there a way to disable

the

silence

detector? I could see that there are several Modes (Fixed,

Adaptive,

etc)

for

it but I can''t figure out where this is initialized in the code.

I may be on the wrong track but I can''t figure out this strange

behavior.

Any help/ideas/suggestions would be greatly appreciated!

Whit

-----Original Message-----

From: opensipstack-devel-bounces at lists dot sourceforge dot net

[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On

Behalf Of

Joegen

E. Baclor

Sent: Monday, April 09, 2007 5:19 AM

To: opensipstack-devel at lists dot sourceforge dot net

Subject: Re: [OpenSIPStack] Audio problems - Jitter and

Comfort Noise

dev at whit dot ca wrote:

Members,

I''m doing some testing with the ATLSIP and opensipstack

libraries

and so

far

with pretty good success. I have written a softphone in C#

using the

samples

provided, however I have a strange issue which I think is

related to

jitter

and/or comfort noise:

Setup:

C# Softphone ----> Asterisk ---> PRI -----> Telco

Once I make a call, the system works fine except if the person

using the

softphone talks for more then about 10-15 seconds (in a row

without

being

interupted). Then, the audio starts to break up and the

person on the

telco

side can''t make out what they are saying. Sometimes this

situation is

reversed

and the person on the softphone can''t make out the person on

the telco

side.

By the way, there aren''t any problems with the telco or

asterisk

setup as

have

SIP hardphones using the system with no problems.

So my question is:

1. Can I send confort noise during silence breaks?

CNG is a codec functionality and is not manually generated by

the

stack.

2. Where can I tweak the jitter-buffer or comfort noise

settings?

Is this

done

SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed

an the

ActiveX properties. Feel free to send in a patch if you get

the chance

to expose it.

in the code itself?

3. Maybe I''m on the wrong track and any suggestions are

welcome!

Look forward to working more with everyone on this exciting

project!

Whit

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Guest

9. Jun 5, 2007 1:03 AM

in response to: Guest

Re: [OpenSIPStack] Cpmfort Noise Support

Hi,

I took time to look at your log. One thing I have noticed is that your
asterisk box keeps on retransmitting the 407 even after it has been
ACKed several times. Would you be able to post your logs to the
asterisk mailing list? From what I can guess, asterisk hates something
that opensipstack sends in ACK. I might have missed something but I''m
sure our ACK is 100% compliant. OpenSIPStack is also sending the
Proxy-Authorization response header in response to the 407.

Joegen

webmaster at dzmeet dot com wrote:

Hi,

This is the log i''m getting

----------------16:09.322----------------

LISTENER STARTED *** 127.0.0.1:5060

----------------16:09.513----------------

LISTENER STARTED *** 192.168.0.53:5060 [*** DEFAULT LISTENER ***]

Thanks

On Sun, 03 Jun 2007 11:26:45 +0800, "Joegen E. Baclor" wrote:

Hi Yacine,

Please send Ilian a level sip 5 log so he can determine the casue and

give you a fix. Thanks.

Joegen

Yacine Auczone wrote:

Hi,

Thanks a lot for all your efforts.

i have succesfully compiled ATLSIP with the new changes, but i have a

little issue now.

i''m not able to make calls since the updates, i''m getting a 403

Forbidden error code whene trying to make a call while i was able to

make calls before.

------------------------------------------------------------------------

Date: Wed, 30 May 2007 17:53:09 +0800

From: ipinzon at solegysystems dot com

To: opensipstack-devel at lists dot sourceforge dot net

Subject: Re: [OpenSIPStack] Cpmfort Noise Support

Hi all,

I have exposed the setting of silence detection mode and audio jitter

delay in ATLSIP and SoftPhoneInterface.

Here are the methods:

DisableSilenceDetection()

- Disables silence detection. Disables CNG as well.

EnableFixedSilenceDetection( ULONG threshold )

- Enables fixed silence detection. Any sound level below the threshold

is treated as silence (and CN is generated as a result). Don''t use too

high threshold values or you''ll only hear comfort noise. Try

threshold=3

as suggested by Whit in another thread.

EnableFixedSilenceDetectionEx( ULONG threshold , ULONG signalDeadband,

ULONG silenceDeadband )

- An extended version of the previous method. Don''t tinker with this

unless you know what you''re doing. For reference on how signalDeadband

and silenceDeadband are used, look in

OpalSilenceDetector::ReceivedPacket().

EnableAdaptiveSilenceDetection( ULONG adaptivePeriod )

- Enables an adaptive silence detection. Supposedly this enables the

threshold to adapt to the current sound level every adaptivePeriod

milliseconds. However, its silence detection doesn''t seem to be very

effective (at least in my machine). I''ll look into this further to see

what''s wrong. This mode with adaptivePeriod=4800 is the default mode

for

ATLSIP.

EnableAdaptiveSilenceDetectionEx( ULONG adaptivePeriod, ULONG

signalDeadband, ULONG silenceDeadband )

- An extended version of the previous method. Don''t tinker with this

unless you know what you''re doing. For reference on how signalDeadband

and silenceDeadband are used, look in

OpalSilenceDetector::ReceivedPacket().

SetAudioJitterDelay( ULONG minDelay, ULONG maxDelay )

- Sets audio jitter delay settings.

Regards,

Ilian

Ilian Jeri C. Pinzon wrote:

Will prioritize this request. This should be available by tomorrow

or on

early Thursday tops.

Regards,

Ilian

Joegen E. Baclor wrote:

Hi Ilian,

Can you provide an ETC for exposing Jitter and Silent Detection

params

in ATLSIP? Seems like a popular request.

Joegen

Ilian Jeri C. Pinzon wrote:

Hi,

We haven''t exposed this yet but we will soon. Please wait for

updates

in this list.

For the meantime, please refer to the attached email on how this

can

be done.

Thanks.

Regards,

Ilian

Yacine Auczone wrote:

Hi All,

First, Thanks a lot for all the great job you are doing for

OpenSipStack and AtlSIP

I''m doing some devlopement test with the Softphone ActiveX, the

quality is very good and no bugs detected, the only thing is

that the

softphone is doing by default some VAD and it is not

transmiting the

silence, so there is no Comfort Noise generation sent whene the

calling party stop talking. i heard about a new ActiveX version

which

will be available and gives the option to enable or disable

CNG, is it

ready? if yes can i have it please?

Other Thing, on my Asterisk Server only G729 Work and not G729A

What''s Wrong ?

------------------------------------------------------------------------

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électroniques sur votre blog ou ajoutez-y des photos, des

blagues et

d''autres infos. C''est gratuit !

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------------------------------------------------------------------------

-------------------------------------------------------------------------

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Download DB2 Express C - the FREE version of DB2 express and take

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------------------------------------------------------------------------

_______________________________________________

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opensipstack-devel at lists dot sourceforge dot net

https://lists.sourceforge.net/lists/listinfo/opensipstack-devel

------------------------------------------------------------------------

No virus found in this incoming message.

Checked by AVG Free Edition. Version: 7.5.472 / Virus Database:

269.8.1/822 - Release Date: 5/28/2007 11:40 AM

------------------------------------------------------------------------

Subject:

[Fwd: Re: [OpenSIPStack] Audio problems - Jitter and Comfort

Noise]

From:

"Joegen E. Baclor"

Date:

Tue, 29 May 2007 18:26:21 +0800

To:

"Ilian Jeri C. Pinzon"

To:

"Ilian Jeri C. Pinzon"

------------------------------------------------------------------------

Subject:

Re: [OpenSIPStack] Audio problems - Jitter and Comfort Noise

From:

"Joegen E. Baclor"

Date:

Thu, 12 Apr 2007 16:40:04 +0800

To:

opensipstack-devel at lists dot sourceforge dot net

To:

opensipstack-devel at lists dot sourceforge dot net

Whit,

Good to hear you nailed it! Can''t wait to see your contributions

you get the chance to expose the other setters/accessors in

ATLSIP.

Whit Thiele wrote:

Joegen,

Thanks for the help. I thought I''d send the list an update on

what

solved my

problem. I changed the Silence Detector to Fixed with a

threshold of

3. This

eliminated all the problems! It seems that the adaptive silence

detector was

constantly incrementing and started affecting things about

10-15 seconds

into a conversation!

I''ll probably put in the ability to change the jitterbuffer and

silence

detector into the ATLSIP library and send this in to the

project in

the next

couple weeks...

Whit

-----Original Message-----

From: opensipstack-devel-bounces at lists dot sourceforge dot net

[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On

Behalf Of

Joegen E. Baclor

Sent: Tuesday, April 10, 2007 9:36 PM

To: opensipstack-devel at lists dot sourceforge dot net

Subject: Re: [OpenSIPStack] Audio problems - Jitter and Comfort

Noise

Whit,

It is probably best to ask this question to

openh323-devel at lists dot sourceforge dot net. However, here''s how to set

the silence detection in code.

OpalSilenceDetector::Param param;

param.Mode = OpalSilenceDetector::NoSilenceDetection;

sfManager.SetSilenceDetectParams( params );

Hope that helps.

dev at whit dot ca wrote:

Joegen,

Thanks for the reply. I''ve been trying different jitterbuffer

settings as

well

as changing the number soundChannelBuffers to a number of

different

settings

which I came across in some online

Opal documentation (

http://www.openh323.org/pipermail/openh323/Week-of-Mon-20051219/076004.html

)

I''ve tried setting the jitter buffer to minimums 25 through to

500

and the

depth

to as high as 15 but nothing is helping. As I described before,

can get

about

10-15 consecutive seconds of decent voice quality and then it

gets very

choppy.

Is anyone else experiencing this?

I am wondering if it may have something to do with the Silence

detection

portion

of Opal. I''ve noticed in the opal.log file that the Silence

Threshold

creeps

upwards the longer the person talks. Is there a way to disable

the

silence

detector? I could see that there are several Modes (Fixed,

Adaptive,

etc)

for

it but I can''t figure out where this is initialized in the code.

I may be on the wrong track but I can''t figure out this strange

behavior.

Any help/ideas/suggestions would be greatly appreciated!

Whit

-----Original Message-----

From: opensipstack-devel-bounces at lists dot sourceforge dot net

[mailto:opensipstack-devel-bounces at lists dot sourceforge dot net] On

Behalf Of

Joegen

E. Baclor

Sent: Monday, April 09, 2007 5:19 AM

To: opensipstack-devel at lists dot sourceforge dot net

Subject: Re: [OpenSIPStack] Audio problems - Jitter and

Comfort Noise

dev at whit dot ca wrote:

Members,

I''m doing some testing with the ATLSIP and opensipstack

libraries

and so

far

with pretty good success. I have written a softphone in C#

using the

samples

provided, however I have a strange issue which I think is

related to

jitter

and/or comfort noise:

Setup:

C# Softphone ----> Asterisk ---> PRI -----> Telco

Once I make a call, the system works fine except if the person

using the

softphone talks for more then about 10-15 seconds (in a row

without

being

interupted). Then, the audio starts to break up and the

person on the

telco

side can''t make out what they are saying. Sometimes this

situation is

reversed

and the person on the softphone can''t make out the person on

the telco

side.

By the way, there aren''t any problems with the telco or

asterisk

setup as

have

SIP hardphones using the system with no problems.

So my question is:

1. Can I send confort noise during silence breaks?

CNG is a codec functionality and is not manually generated by

the

stack.

2. Where can I tweak the jitter-buffer or comfort noise

settings?

Is this

done

SoftPhoneManager::SetAudioJitterDelay(). It is not yet exposed

an the

ActiveX properties. Feel free to send in a patch if you get

the chance

to expose it.

in the code itself?

3. Maybe I''m on the wrong track and any suggestions are

welcome!

Look forward to working more with everyone on this exciting

project!

Whit

-------------------------------------------------------------------------

Take Surveys. Earn Cash. Influence the Future of IT

Join SourceForge.net''s Techsay panel and you''ll get the

chance to

your

opinions on IT & business topics through brief surveys-and

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May 29, 2007 5:29 AM

[OpenSIPStack] Cpmfort Noise Support

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