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? Thanks in advance. Best regards, Cerrajetto. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
