I'm trying to pass 4 phone calls over a 64Kbps link, i´m using frf.12 fragmentation and cRTP. Theoretically each call consumes about 11kbps then i would be able to pass 4 call by the link. Doing the tests i can only pass 2 call without problem and as soos as i dial the third call the other calls begin to fail and the packet drops increases.
My question, is the 11 Kbps just one way and i have to consider twice that bandwith?
Your serial link is full duplex so the bandwidth in the Cisco charts is total needed for a two-way call. 11.2 kbps is fine for a call. This assumes you are using the default payload of 20 bytes. If you are using the alternative 40 byte payload your required bandwidth drops to 9.6 kbps.
Just to make certain that we have not overlooked the obvious (of which I have been guilty too many times in the past).
What are you using for compression? Check the codec on each router with a show voice dial-p. Your assumed bandwidth is for G.729 20 byte payload with FRF 12 and cRTP (without VAD).
If you are using 7960 IP phones pressing the i key twice (during a call) will show the transmit and receive codecs.
Next check to insure that you are really running cRTP. The command when running frame relay is not ip rtp header-compress. It is frame-relay ip rtp header-compress and the corresponding show command is show frame-relay ip rtp header-compress.
I assume you not running any data on this link during your tests!
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