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