Re: [OpenSIPStack] Open Sip Stack / IVR Features
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Re: [OpenSIPStack] Open Sip Stack / IVR Features
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From: Christian W. <cwa...@gm...> - 2009-01-08 15:39:07
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Hi Joegen, thanks for your reply. To your first question, I don't know exactly what are the differences There ... I need something which handles the protocol on a single or multiple line/s. If audio data is received it should be automatically converted to PCM (inband dtmf tone detection) which also makes mixing easier. A dream would be to have a UAC and a UAS running (so a soft phone can be connected to the stack (for call center applications)). To the conferencing feature: I need add hoc conferences where I can decide if - a user can speak / listen - record only the user or the whole conference - play on user side or in the whole conference Dialout = INVITE / Call a user / number To the delay: In the past I tried to send g711 alaw sounds with a buffer size of 2048 Bytes (250 ms). The stack was connected to an asterisk PBX where a soft phone was logged in... For incoming RTP data (which comes from the soft phone everything works since the delay comes from the microphone / audio device) but for sending audio data I dont know how to calculate the delay, I thought someone is buffering the data and the stack will take care delivering the next block to the soft phone (I thought the send media function would block or fail until the Buffer on the other side is played). I anyone can help me maybe I could ask my boss if he can "donate" some Money for the project / person since I really need this functionality (or I need to use the Office Live communications server which I dont want) the only missing criteria to use my software is SIP. I have an ISDN based IVR which can do everything we need except SIP (conferencing, dtmf, database support, dll plugin support, snmp monitoring, ...) which runs on CAPI hardware here on 4 E1 with a load of 10 % if the debugger of the IVR is not connected to the runtime ... So maybe someone can help me ... Kind regards Christian -----Ursprüngliche Nachricht----- Von: jo...@op... [mailto:joe...@gm...] Gesendet: Mittwoch, 7. Januar 2009 16:55 An: ope...@li... Betreff: Re: [OpenSIPStack] Open Sip Stack / IVR Features Christian Wallukat wrote: > Hi all, > > > long time ago I invested some time in the stack to develop a SIP based IVR. > In the past the problem was playing audio data on the client side (the file > was played to fast). > > So I want to ask if there is something built in which can do: > > - Media conversion to PCM > Are you referring to Media Server Module, Softphone Module or B2BUA Module? MS can record to PCM if the codec used is supported. This includes conversion from G.729/G.723.1 to PCM if Voice Age G.729 or Sipro G.723.1 libraries are detected by opensipstack configure script. All open source codecs are supported. PCMA PCMU G.726 GSM Speex iLBC > - Conferencing > Mixing is not supported yet. Are you asking for an MCU/Conference server feature or an Ad Hoc Conference UA functionality? > Now I have some time again and I wanna ask if there is now something > In the stack which can do: > > - Play / Record > MS has this functionality > - DTMF Tone recognition / generation > RFC 2833. > - Conferencing > This requires mixing and is not yet supported in any of the three modules. > - Dialout > > Sorry what's dialout? UAC functionality? > I think I now got the problem I had in the past: > > Since I play a file, the stack would send the data as I deliver it... > So if this is still the same, how to time the data on my side so they > a) would not be lost and b) played correctly on the other side? > You need to delay sending based on the media format. Different formats have their own specific algorithmic delays. What format are trying to play exactly? G.729 for example needs an algorithmic delay of 10 ms per frame. > > Kind regards > > > Christian > > > > > ---------------------------------------------------------------------------- -- > Check out the new SourceForge.net Marketplace. > It is the best place to buy or sell services for > just about anything Open Source. > http://p.sf.net/sfu/Xq1LFB > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------ > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.176 / Virus Database: 270.10.3/1879 - Release Date: 1/6/2009 5:16 PM > > ---------------------------------------------------------------------------- -- Check out the new SourceForge.net Marketplace. It is the best place to buy or sell services for just about anything Open Source. http://p.sf.net/sfu/Xq1LFB _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.176 / Virus Database: 270.10.4/1880 - Release Date: 07.01.2009 08:49 |
