Hi there, up till now I had this two-box setup in mind:
* no.1: public IP * no.2: private IP, registers with no.1, serves a small office with clients behind NAT See we'd get something like this: SIP client (GSM) --> *1 --> IAX2 (iLBC) --> *2 --> G.711 --> MGCP UA The codec of the SIP client (on the Internet) I don't have full control over, that depends on the capabilities of the client, so it can be GSM (preferably) or something else. iLBC appears to be great for inter-* connections when bandwidth is an issues (from what I read). G.711 finally is required since that is the only common protocol between * and the IP phone available. But then I stumbled across the passage I quoted below. Should I reconsider the setup to at least remove one of the transcodings? Or is the document's author simply wrong? Greetings, Philipp 2) Transcoding: To be avoided at all times Transcoding is the conversion of a voice stream with one codec to a voice stream with another codec (e.g. G.729 to G.7.23). Transcoding dramatically degrades the voice quality. It has to be avoided at all times. Comment: Stay with G.711 until the cost of bandwidth becomes an issue, then stick to one choice of your trade-off decision. The above was taken from: http://www.beltug.be/pages/Pdfs/Checklist_VoIP.pdf _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
