or audio quality. If it's cost savings, you could push 25 calls through a T1 using GSM encoding, but it would not sound quite the same as a regular line. If you use G.711 (mu/A-Law) then you would get toll quality audio but only be able to push about 17 calls through at once...
based on my albeit rough benchmarks, a T1 (1.5Mbps) should be able to handle about 46 calls with GSM and about 20 calls with ulaw. i just did a quick test with a single call as follows:
SIP/ulaw IAX/gsm WellGate 3502A <--------> rl0 Asterisk rl1 <-------> IAXTel <--- IVR
# netstat -w 10 -I rl1 -d # sample at 10-second intervals
input (rl1) output
packets errs bytes packets errs bytes colls drops
509 0 40151 262 0 28768 0 0
504 0 39759 254 0 28274 0 0
506 0 39860 256 0 28324 0 0
502 0 39571 254 0 28224 0 0(this gives about 32kbps per direction for GSM, IP overhead inclusive)
# netstat -w 10 -I rl0 -d # sample at 10-second intervals
input (rl0) output
packets errs bytes packets errs bytes colls drops
250 0 93500 508 0 108712 0 0
250 0 93500 501 0 107214 0 0
250 0 93500 497 0 106358 0 0(this gives about 87kbps per direction for ulaw, IP overhead inclusive)
i wasn't doing much talking, so the input figures from rl1 and the output figures from rl0 should be used. they match john todd's observation on the asterisk wiki.
without IAX trunking and making worse case assumptions on IP overhead, the above figures would be good. IAX trunking would save somewhat on IP overhead as per http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2
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