Re: [OpenSIPStack] Open Sip Stack / IVR Features

[<jo...@op...>]

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
>
>
>
>
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