Hello, sorry for my late reply: @mazilo: I have not tested the HRTF my own but I know that it is not such an extreme thing. The point is: 5 people in a conference => every participant needs 4 other people mixed up => 5x4 = 20 HRTF mixings parallel => n*(n-1) complexity :-(. If I integrate this stuff in FreeSWITCH I will, of course, offer it as open source as well (we can try to integrate it or whatever the main developers want to have). But at first I need to get it working. @Steve: Thanks. Is there some documentation for which I am too stupid to find it? @Anthony: - Playing vs. Codec: Sounds logical! Codec: What about L16. I think there is a 1 and a 2 channel version available: http://en.wikipedia.org/wiki/RTP_audio_video_profile - I completely understand why there is no stereo until now. - Portaudio: I'm not sure if I really need this mod for my purpose. It was only a nice idea to test and debug how media is handled. - I understood SIP the way as it is only a protocol for "managing" things. The media flow uses RTP and RTP can use different codecs and different numbers of channels. The SDP can also handle different codecs for sending and receiving. Therefore, I thought, sending mono and receiving stereo is completely SIP regular!?? @tim: I'm not sure if I understood your idea completely in the right way: Your main idea is to use one sip session for each channel? But mixing would still be done by the server, not the client? How should I take care that the left and the right channels, processed by two different SIP + RTP sessions, are played simultaneously? Thanks to all your great support!!! MatzeMuc86 -----Ursprüngliche Nachricht----- Von: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Tim St. Pierre Gesendet: Freitag, 25. Februar 2011 19:59 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] implementing stereo (e.g. portaudio to test it easily) Here's an idea - it might be a little easier to make interoperate with other SIP devices. What if you built a conferencing endpoint that mixed all the audio locally into a 3D matrix? Each participant in the call could use any of the common codecs, but would have an extra SIP header than indicated their position in the sound field. Here's a bit of an overview: A regular Freeswitch system takes care of registering endpoints, routing, etc. Just like a normal PBX. The only thing special it would do is set add a custom SIP header with a person's sound field position, based on a directory variable. Each conference unit would be a stripped down freeswitch system with a custom modified mod_conference. In addition to mixing audio to mono like the regular mod_conference does, it would also generate a stereo mix, based on the position information of each participant. It could dump out the stereo audio directly to a named pipe, mod_shout, or directly to audio hardware. You could probably shoehorn a stereo input from hardware as well. Essentially, you are building a stereo conference phone. If you want to join up multiple conference units, just establish two sessions between each conference - one panned hard left, the other hard right. They can then use G722 without any modification. This probably isn't quite as nice as having true stereo codecs and media handling, but it may be easier to implement. After all, telecom is very much a mono world. -Tim
The main reason we do not have stereo completely implemented is there was not really any codec or devices that could handle it to bother testing with.
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