> On Apr 8, 2005 12:22 PM, snacktime <[EMAIL PROTECTED]> wrote: > > So far it seems that the major thing affecting voice quality on my * > > box is codec translation. How much cpu is required to translate even > > a single channel without getting static like sounds or other obvious > > translation issues? I know this probably depends on the codecs > > involved, but are there any general guidelines to follow?
If you're talking about a relatively small * system, codec translation is not a big deal. From the * CLI, do a "show translation" to get a rough idea what kind of consumption happens between various codecs. > One more question. I've been trying to figure out the best > combination of codecs to use. So far it seems that g729 is the low > bandwidth codec most widely supported. gsm seems to be supported by > providers but not by sip devices. g726 the opposite. I'm thinking it > might be worth it to just pay digium to license g729 and record all > our own voice prompts. Having the g729 license will enable us to > record files in g729 format correct? Depends on what you're trying to do with asterisk. If its a dedicated corporate pbx with no interfaces to the internet, the codec selection is likely to be governed by what the sip phones can support. If you're trying to use * as a soho system, its likely governed by a combination of what your itsp will support and what phones you are using, combined with available braodband bandwidth. You'll find most of these items mentioned/discussed in the wiki. Take a look. Most popular codecs are g711, g729, and gsm (for iax). For the small cost of a g729 license, dump five or ten into your system and add to them later if actually needed. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
