Re: [Opalvoip-user] best way for sending audio (wav) files
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From: liviu u. <liv...@ya...> - 2012-10-11 06:50:24
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Ok, I've rebuild the libraries with the expat tool to include the IVR end point and so far seems to work, even more, it's exactly what I need (thanks a lot!). In this case, is it ok to still have a PCSSEndPoint attached to the manager? Right now I'm only using the PCSSEndPoint::AcceptIncomingConnection(const PString &) when OnShowIncoming in PCSSEndPoint occurs. Thanks in advance, liviu --- On Tue, 10/2/12, Robert Jongbloed <ro...@vo...> wrote: From: Robert Jongbloed <ro...@vo...> Subject: Re: [Opalvoip-user] best way for sending audio (wav) files To: opa...@li... Date: Tuesday, October 2, 2012, 11:24 AM That depends on the application. For example if all you want to do is answer a call, play a message, then hang up, then using PCSS endpoint should be fine. But if you want to do complicated stuff, stitch together a sequence of WAV files, wait for answer, stitch together another set, etc. Then that is what IVR endpoint is for. Pretty sure there is logging for this. Robert Jongbloed OPAL/OpenH323/PTLib Architect and Co-founder. Commercial support at http://www.voxlucida.com.au On 2/10/2012 5:13 PM, liviu ursulescu wrote: Is it important when to switch to the wav file? When I create an instance of PCSS endpoint, it will take the "default" device. Also, Opal manager will create the necesary connection (pcss and sip) for me too, when a call arives. So do I need to switch in a certain connection phase? Can the PSoundChannel function be involved somehow to allow to replay wavs so the call doesn't hang up? Or they act only as local players/recorders functions. And, for debugging purposes, in the opal trace, is there any info that the wav is actually send/played? Thanks & regards, Liviu --- On Thu, 9/27/12, Robert Jongbloed <ro...@vo...> wrote: From: Robert Jongbloed <ro...@vo...> Subject: Re: [Opalvoip-user] best way for sending audio (wav) files To: opa...@li... Date: Thursday, September 27, 2012, 11:16 AM Yes, it can be the name of a WAV file. The PSoundChannel functions or for playing WAV files are for playing to the sound channel itself, that is on the PC speaker. What you want is the "fake" sound channel that isn't a physical sound card, but the WAV file. And that is sort of automatic if you just use the WAV file name as the "device" name. But the issue, as I said earlier, is it is hard to handle what happens when the WAV file ends. There is a good chance the call will just hang up. Robert Jongbloed OPAL/OpenH323/PTLib Architect and Co-founder. Commercial support at http://www.voxlucida.com.au On 27/09/2012 4:29 PM, liviu ursulescu wrote: Ok, for the first case I suppose that the function OpalPCSSEndPoint::SetSoundChannelPlayDevice(const PString &name) is involved somehow. If this is true, can the parameter "name" specify a soundchannel or even the wav file that needs to be played? I see that the PSoundChannel has some mechanisms on how to deal with a sound files, and was wondering if it can be used to play/stop some already recorded messages. Thank you in advance, Liviu --- On Tue, 9/25/12, Robert Jongbloed <ro...@vo...> wrote: From: Robert Jongbloed <ro...@vo...> Subject: Re: [Opalvoip-user] best way for sending audio (wav) files To: opa...@li... Date: Tuesday, September 25, 2012, 1:42 PM Maybe I should put this into the FAQ, seems to be asked a lot lately! There are two mechanisms: You can switch the audio device in PCSSConnection from a physical sound card to the WAV player device. The problem with this mechanism it is not easy to deal with what happens when the WAV file finishes. It tends to close the media stream down, so only good if you are planning to play message and hang up. The second mechanism is to use the IVR subsystem. See opal/samples/openphone for how to do that. Robert Jongbloed OPAL/OpenH323/PTLib Architect and Co-founder. Commercial support at http://www.voxlucida.com.au On 25/09/2012 4:23 PM, liviu ursulescu wrote: Hello everybody, I'm currently working at a project which is something similar to an answering machine, and uses the ISDN lines to send the voice to the caller (according to the CAPI protocol). I want to extend it to work with VoIP to. I've choose OPAL for this, and so far managed to "call" my app, from an internal SIP PBX - and using the 3CX voip phone app., by extending the OpalManager, SIPEndpoint, and PCSSEndpoint. Now my question is, how is best to send the audio messages? Can I attach somehow my wav files to the existent OpalMediaStream created for each connection, to send them over RTP, or do I need to create another endpoint? (perhaps a localEndpoint with the appropriate connection, and RTP session). the format is PCMA so i'm thinking that there is no problem with the codecs.. 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