Q1 In the scenario, where you have all incoming PSTN calls coming into 1 site and then being routed across the wan
(G.729) to the relevant user.
How would the transcoding work in this scenario? Usually you would terminate a TDM with a G.711 packet.
but as you would be going across the wan would it automatically make it G.729 without using extra DSP resources?
If then the call was forwarded to voicemail ( back over the wan to the main site) Would the same DSP resource be
used to convert to G.711?
I think in Summary what I am saying is, am I right in assuming that only one DSP channel would be used for
the whole duration of the call. NO extra DSP resources are required? No configuration required?
Q2. With a Vmail server located at a main site and 2 users at a remote site. When these 2 users call eachother,
they would use G.711 as they are on the same site. The call goes to Vmail, which has to go over the wan (G.729)
and then to communicate with Unity, which uses G.711.
What/how many transcoders would be used and what config required?
I am assuming each Voice Gateway would do the transcoding, but not sure how the scenario plays out.
TDM is not G711, is just PCM Audio,
When it is being converted to Voice packets its now G711 or G729 depending the codec being used.
Also for your question the DSP may use more resources (not transcoding) when doing a G729 call or Resources when trancoding
Readthe voice termination section.
Q2. If your VM server doesnt support G729, you will need 1 Transcoder
IP Phone -----G729------- VM
--G729--XCODER G711-- VM
Last document is really good for calculating resources as well.