>>> As a podcaster I use Asterisk extensively and often have several people >>> in >>> a conference room. We'll record the calls via a SIP phone connected to >>> a >>> sound mixer. Is there an easy way to bump up the audio bitrate for all >>> callers connected to the Asterisk server and improve the general sound >>> quality? The server is not used much outside of recording the podcast. >>> We're not opposed to compiling Asterisk ourselves to get the results >>> we'd >>> like. >> >> Let me understand first: the SIP phone doing the recording is not one >> of >> the people on the conference? It's in >> monitor mode, for recording purposes only? >> >> If that's the case, then you can't achieve audio quality higher than the >> individual conference node channels >> themselves -- sort of a 'lowest common denominator' situation. If you >> could get all nodes using a wideband codec (say >> G722), and if Asterisk supports wideband mixing and recording (i.e. >> everything done at 16 kHz sampling rate), then you >> might be able to do it. >> >> -Jeff >> > > Jeff, > So the first thing to improve audio quality is to switch over to a higher > quality codec like G722. What are the other higher quality codecs we can > use? Everyone connecting should make sure they're using the higher quality > codec? Is there any way to configure a stock Asterisk install to use > wideband mixing or will we have to compile our own? > Thanks again > > Pat > > >
I found this link: http://www.voip-info.org/wiki/view/Asterisk+codecs So every client that connects to the conference would have to be configured to use whatever codec we wind up using. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
