Hi. I'm interested in using the sflphone software to record a podcast with the assistance of an asterisk server. The highest quality codec supported by asterisk bridging is 32kHz speex. I've noticed though that the quality is somewhat bad. I've seen options to modify the encoding quality level for speex before, and I see references to quality levels in the sflphone source code, but I can't find them in the sflphone user interface.
Can anyone give me any advice in improving the VoIP audio quality to its maximum levels with this software? _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
