opensipstack-devel

[OpenSIPStack] [ATLSIP] Ringing Sounds

[Whit T. <de...@wh...>]

Hey guys,

Where and when should the ringing sounds be generated? I use Asterisk so
when a call is launched, asterisk generates the ringing sound. If I don't
disable the PlayRingingSound methods, I get "double" rings. 


Should this be a configurable setting in the ATLSIP library?

Whit

Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

[Joegen E. B. <joe...@gm...>]

This may happen if your gateway sends a 180 without SDP followed by 180 
or a 183 with SDP.    This can be corrected by stopping the false ring 
in OnProgress()

void SoftPhoneSIPEndPoint::OnProgress(
  CallSession & session,
  const SIPMessage & alerting
)
{
  OpalOSSEndPoint::OnProgress( session, alerting );
  PString info = session.GetTargetURI().AsString();
  if( session.GetType() == CallSession::Client )
    m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const 
char *)info );
}

Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in 
this method as a quick hack.   I think a cleaner way of doing this is to 
not honor early media at all and retain the false ring if the call has 
already received a no-media Alerting packet prior to the 183.  perhaps 
we can set this in the stack level.   I am open to suggestions. 

What's your view Ilian?

Joegen

Whit Thiele wrote:
> Hey guys,
>
> Where and when should the ringing sounds be generated? I use Asterisk so
> when a call is launched, asterisk generates the ringing sound. If I don't
> disable the PlayRingingSound methods, I get "double" rings. 
>
>
> Should this be a configurable setting in the ATLSIP library?
>
> Whit
>
>
>
> -------------------------------------------------------------------------
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>
>
>   

Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

[Ilian J. C. P. <ip...@so...>]

RFC 3960 *suggests* the following policy:

=========================================================
1. Unless a 180 (Ringing) response is received, never generate
local ringing.

2. If a 180 (Ringing) has been received but there are no incoming
media packets, generate local ringing.

3. If a 180 (Ringing) has been received and there are incoming
media packets, play them and do not generate local ringing.

...

That is, any UA should play incoming media packets (*and stop local 
ringing tone
generation if it was being performed*) in order to avoid media clipping,
even if the 200 (OK) response has not arrived.
=========================================================

I think interrupting the local ring tone, if it is playing, in favor of 
the early media
is acceptable already.

- Ilian

Joegen E. Baclor wrote:
> This may happen if your gateway sends a 180 without SDP followed by 180 
> or a 183 with SDP.    This can be corrected by stopping the false ring 
> in OnProgress()
>
> void SoftPhoneSIPEndPoint::OnProgress(
>   CallSession & session,
>   const SIPMessage & alerting
> )
> {
>   OpalOSSEndPoint::OnProgress( session, alerting );
>   PString info = session.GetTargetURI().AsString();
>   if( session.GetType() == CallSession::Client )
>     m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const 
> char *)info );
> }
>
> Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in 
> this method as a quick hack.   I think a cleaner way of doing this is to 
> not honor early media at all and retain the false ring if the call has 
> already received a no-media Alerting packet prior to the 183.  perhaps 
> we can set this in the stack level.   I am open to suggestions. 
>
> What's your view Ilian?
>
> Joegen
>
> Whit Thiele wrote:
>   
>> Hey guys,
>>
>> Where and when should the ringing sounds be generated? I use Asterisk so
>> when a call is launched, asterisk generates the ringing sound. If I don't
>> disable the PlayRingingSound methods, I get "double" rings. 
>>
>>
>> Should this be a configurable setting in the ATLSIP library?
>>
>> Whit
>>
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems?  Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >> http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>   
>>     
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
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> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>   

Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

[H.Kropf <mai...@gl...>]

This method does not work, because after it the method 1 is caused and
the sound does not interrupt. It can be necessary to add a variable 123
in class CallSession and to appropriate to it value TRUE in a method 1,
and in a method 2 to read out it and to appropriate to it value FALSE

This "quick hack" method does not work, because after

void SoftPhoneSIPEndPoint::OnProgress(...)
{
  ... 
  m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); 
  ...
}

is called

void SoftPhoneSIPEndPoint::OnAlert(...)
{
  ... 
  m_Manager.GetSoftPhoneInterface()->PlayRingBackTone(); 
  ...
}

and the RingBackTone does not interrupt


It can be necessary (for example) to add a boolean variable m_NoRingBack 
in class CallSession and to use it so:

void SoftPhoneSIPEndPoint::OnProgress(
  CallSession & session,
  const SIPMessage & alerting
)
{
  ...
  if( session.GetType() == CallSession::Client )
    session->m_NoRingBack = true;
  ...
}

void SoftPhoneSIPEndPoint::OnAlert(
  CallSession & session,
  const SIPMessage & alerting
)
{
  ...
  if(!session->m_NoRingBack) 
    m_Manager.GetSoftPhoneInterface()->PlayRingBackTone();

  session->m_NoRingBack = false;
 ...
}



Joegen E. Baclor wrote:
> This may happen if your gateway sends a 180 without SDP followed by 180 
> or a 183 with SDP.    This can be corrected by stopping the false ring 
> in OnProgress()
>
> void SoftPhoneSIPEndPoint::OnProgress(
>   CallSession & session,
>   const SIPMessage & alerting
> )
> {
>   OpalOSSEndPoint::OnProgress( session, alerting );
>   PString info = session.GetTargetURI().AsString();
>   if( session.GetType() == CallSession::Client )
>     m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const 
> char *)info );
> }
>
> Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in 
> this method as a quick hack.   I think a cleaner way of doing this is to 
> not honor early media at all and retain the false ring if the call has 
> already received a no-media Alerting packet prior to the 183.  perhaps 
> we can set this in the stack level.   I am open to suggestions. 
>
> What's your view Ilian?
>
> Joegen
>
> Whit Thiele wrote:
>   
>> Hey guys,
>>
>> Where and when should the ringing sounds be generated? I use Asterisk so
>> when a call is launched, asterisk generates the ringing sound. If I don't
>> disable the PlayRingingSound methods, I get "double" rings. 
>>
>>
>> Should this be a configurable setting in the ATLSIP library?
>>
>> Whit
>>
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems?  Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >> http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>   
>>     
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
> opensipstack-devel mailing list
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> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>   

Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

[H.Kropf <mai...@gl...>]

Sorry

This "quick hack" method does not work, because after

void SoftPhoneSIPEndPoint::OnProgress(...)
{
  ...
  m_Manager.GetSoftPhoneInterface()->StopRingBackTone();
  ...
}

is called

void SoftPhoneSIPEndPoint::OnAlert(...)
{
  ...
  m_Manager.GetSoftPhoneInterface()->PlayRingBackTone();
  ...
}

and the RingBackTone does not interrupt


It can be necessary (for example) to add a boolean variable m_NoRingBack
in class CallSession and to use it so:

void SoftPhoneSIPEndPoint::OnProgress(
  CallSession & session,
  const SIPMessage & alerting
)
{
  ...
  if( session.GetType() == CallSession::Client )
    session->m_NoRingBack = true;
  ...
}

void SoftPhoneSIPEndPoint::OnAlert(
  CallSession & session,
  const SIPMessage & alerting
)
{
  ...
  if(!session->m_NoRingBack)
    m_Manager.GetSoftPhoneInterface()->PlayRingBackTone();

  session->m_NoRingBack = false;
...
}



Joegen E. Baclor wrote:
> This may happen if your gateway sends a 180 without SDP followed by 180 
> or a 183 with SDP.    This can be corrected by stopping the false ring 
> in OnProgress()
>
> void SoftPhoneSIPEndPoint::OnProgress(
>   CallSession & session,
>   const SIPMessage & alerting
> )
> {
>   OpalOSSEndPoint::OnProgress( session, alerting );
>   PString info = session.GetTargetURI().AsString();
>   if( session.GetType() == CallSession::Client )
>     m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const 
> char *)info );
> }
>
> Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in 
> this method as a quick hack.   I think a cleaner way of doing this is to 
> not honor early media at all and retain the false ring if the call has 
> already received a no-media Alerting packet prior to the 183.  perhaps 
> we can set this in the stack level.   I am open to suggestions. 
>
> What's your view Ilian?
>
> Joegen
>
> Whit Thiele wrote:
>   
>> Hey guys,
>>
>> Where and when should the ringing sounds be generated? I use Asterisk so
>> when a call is launched, asterisk generates the ringing sound. If I don't
>> disable the PlayRingingSound methods, I get "double" rings. 
>>
>>
>> Should this be a configurable setting in the ATLSIP library?
>>
>> Whit
>>
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems?  Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >> http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>   
>>     
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>