Yes, but my problem is that I did exactly that, I was switching the codec to Speex in OpenMeetings and tested the recording, the problem was that FFMPEG was not able to strip the audio correctly from the stream.
However I think there are also some changes in your code required to transcode Speex2GSM and GSM2Speex. The 8KHz sample rate is a strict requirement? I think 8KHz is rather bad quality, is it possible to use higher sampling rate? Maybe 16 or 32 KHz? Sebastian 2012/2/22 Тимур Тлеукенов <[email protected]> > Hi Sebastian, > > It uses NellyMoser and g.711 U-law. Stream from openmeetings encoded with > NellyMozer, SIP transport transcodes its to g.711. > g.711 packets from SIP-server are transferred to red5 without any changes. > > Currently it supports only NellyMozer codec with 8KHz sampling frequency, > so in openmeetings config.xml microphoneRateBest must be set to value 8. > > Codec g.711 U-law has been selected because it is supported by both flash > player and Asterisk. > > ActionSctipt 3 gives ability to specify codec, so when we switch > openmeetings to new video component, NellyMozer codec will not be required > at all. > > On Wed, Feb 22, 2012 at 10:54 PM, [email protected] < > [email protected]> wrote: > > Hi Timur, > > > > your SIP integration implementation, does it require to configure a > special > > Codec for the Audio in the Flash Stream? > > It does work currently with NellyMoser right? > > > > Sebastian > > > > -- > > Sebastian Wagner > > http://www.openmeetings.de > > http://incubator.apache.org/openmeetings/ > > http://www.webbase-design.de > > http://www.wagner-sebastian.com > > [email protected] > > > > -- > Timur Tleukenov > > -- Sebastian Wagner http://www.openmeetings.de http://incubator.apache.org/openmeetings/ http://www.webbase-design.de http://www.wagner-sebastian.com [email protected]
