> Hello: > > I would like to implement a VoIP telephony system; client's > software/hardware only supports SIP with the g.711 codec. > > It's very huge in wan communications (about 110 Kbps) !!! > > I'm looking for systems that could reduce this bandwidth, at this time I > can't change the client side. > > Can Asterisk act as a gateway to convert g.711 to GSM, and then reduce the > wan traffic?. The idea is on-the-fly conversion: > > client 1 -> Asterisk GW 1 -> wan -> Asterisk GW 2 -> client 2 > > client 1 calls client 2 and sends g.711 > Asterisk GW 1 converts on-the-fly g.711 to GSM > Asterisk GW 2 converts on-the-fly GSM to g.711 > client 2 receives g.711 from client 1 > (and vice-versa) > > Is it possible to do it with Asterisk? >
YES! > Thanks in advance. > > Best regards, > Cerrajetto. > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon: +49 30 79705390 fax: +49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
