Very basic question , but I cant seem to see why a full DSP resource is required
when sending voip calls over a Digital trunk.
Assuming that both sides use G.711 and no transcoding is required what role does the DSP perform ?
Is it just to add the IP/UDP/RTP Packet headers ?
I mean no sampling / quantaziation or encoding is being done...
If so, cannot this be done just in s/w ?
(Perhaps the obvious answer is no, and that is why a DSP is required)
Or am I missing somethig more ?
Both G711 and T1's use 64kb/sec uncompressed formatting. This boils down to Nyquist's theorem with the maximum sample size for the frequency range. It's estimated that human voice can reach 4000 Hz so we sample the voice at 8000 Hz (forgive me if these numbers are a bit off).
The DSPs also do a great number of other things. The echo cancellation, creating of playout delay buffers, input gains, output attenuations, DTMF detection, and other tone detection (fax/modem) is all done through the DSP. If you use SIP for instance, and the gateway receives a RFC 2833 DTMF packet, something on the router must generate the DTMF tone in the circuit switched world. The DSP in this case is what is used to change a RFC 2833 packet into DTMF.
The ulaw/alaw is normally done on the TDM side as well as the IP side, so the DSPs aren't generating this as much as they are normally passing it on (but they can transcode this as well).
The RTP packet headers are going to be built in IOS, but the payload of the packets is going to be processed in the DSP as long as there is an RTP packet that needs to be built.