Re: [OpenSIPStack] [ATLSIP] Ringing Sounds
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joegenbaclor
From: Whit T. <de...@wh...> - 2007-10-25 19:05:16
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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 > > > ------------------------------------------------------------------------- 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 |