opensipstack-devel

Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

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

Illian,

Thanks for your information! In my particular instance, it was easier to
temporarily disable the local ringing sound because I think my pbx
(asterisk) isn't sending the correct packets. I'm investigating this further
before I re-enable the local ringing sounds. I'll let you know what I find
out!

Whit


-----Original Message-----
From: ope...@li...
[mailto:ope...@li...] On Behalf Of Ilian
Jeri C. Pinzon
Sent: Wednesday, October 24, 2007 6:35 AM
To: jb...@so...; ope...@li...
Subject: Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

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
>>
>>
>>
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>> ope...@li...
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>>
>>
>>   
>>     
>
>
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>
>   


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Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

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

Hi Whit,

No problem with that. Just keep in mind that you will lose ringing 
totally when calling UAs that don't generate their own ringing tones.

Thanks.

- Ilian

Whit Thiele wrote:
> Illian,
>
> Thanks for your information! In my particular instance, it was easier to
> temporarily disable the local ringing sound because I think my pbx
> (asterisk) isn't sending the correct packets. I'm investigating this further
> before I re-enable the local ringing sounds. I'll let you know what I find
> out!
>
> Whit
>
>
> -----Original Message-----
> From: ope...@li...
> [mailto:ope...@li...] On Behalf Of Ilian
> Jeri C. Pinzon
> Sent: Wednesday, October 24, 2007 6:35 AM
> To: jb...@so...; ope...@li...
> Subject: Re: [OpenSIPStack] [ATLSIP] Ringing Sounds
>
> 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/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>   
>>     
>
>
> -------------------------------------------------------------------------
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> Still grepping through log files to find problems?  Stop.
> Now Search log events and configuration files using AJAX and a browser.
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> _______________________________________________
> 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
>
>
>