JackTheTech wrote:Not really voip now days runs on a G729 codec which uses bugger all bandwidth
While G.729 (not going to get bogged down in the variants) only uses 8 kbit, when planning VoIP bandwidth requirements you need to take the whole session into consideration.
So G.729 at a standard sample rate of 10ms and two samples per packet is sized at 20 bytes. 50 pps = 8000 bps.
So then we need to incorporate the RTP header of 12 bytes, the UDP header of 8 bytes and the IP header of 20 bytes for a total of 40 bytes.
Then we need to consider the actually medium that its travelling over. Standard ethernet has a header of 18 bytes.
So all up we have total of 78 bytes @ 50 pps = 31200 bps.
For G.711 which everyone should be using at home (why sell yourself short on an ADSL connection?) its 160 bytes + 58 bytes @ 50 pps = 87200 bps
Keeping in mind that this doesn't include RTCP or the actual call control protocol (H.323/SIP/MGCP/SCCP) requirements.
Blacky wrote:You dont need a PABX as such, all you'd need is a call manager....ie a computer running software with a gateway
http://www.asterisk.org/for example
Call-manager typically refers to the Cisco product running MGCP, H.323, SIP and SCCP protocols, SCCP being proprietary and until CUCM v6, SIP functionality was very limited.
Asterisk is a PBX product taking the role of a SIP Proxy and Registrar (SIP being built as a p2p protocol) which also supports MGCP and H.323