On Mon, 2003-03-10 at 06:04, Cerrajetto wrote: > 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?
Not only is it possible, but you can convert from SIP to IAX and gain the ability to go behind a NAT firewall with ease. Client1 G.711 SIP Asterisk SIP receipt Asterisk IAX GSM transmit WAN Asterisk IAX GSM receipt Asterisk SIP g.711 transmit Client 2 Similar to what I used at home with a ATA186, ata to asterisk via G711 SIP, IAX from my asterisk machine to the office using GSM, then offload to PSTN. -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
